| 00:05.21 | hads | Thanks file, you fixed a bug that was bugging me :) |
| 00:06.44 | Strom_C | did you bug the bug so you could discover what was being said in the secret "bug hads" meetings? |
| 00:07.44 | hads | Wow, I had to read that sentance about 3 times. Stupid mondays. |
| 00:09.35 | partition | hads: which one? |
| 00:11.09 | hads | When doing a *8 pickup I was getting distorted audio and the svn up I did this morning fixed it. |
| 00:11.17 | partition | ah yes |
| 00:11.26 | hads | Yay you :) |
| 00:11.43 | partition | I know why exactly it did that, and where |
| 00:12.00 | partition | but I won't dare touch it anymore :D |
| 00:12.05 | hads | heh |
| 00:13.04 | trevarthan | so..... what's the next best free low bandwidth codec beside iLBC? |
| 00:13.36 | trevarthan | I'm looking at the Linksys SPA 942 sip phone, and it doesn't list iLBC as a codec... I'm wondering if that will be a problem in the future... |
| 00:14.54 | trevarthan | I had relatively good results with iLBC in the past. Is iLBC not a standard? |
| 00:15.08 | Strom_C | everyone tends to support g.711 and g.729 |
| 00:15.08 | [1]qrao | trevarthan, g.726 is a pretty good LBC |
| 00:15.41 | Strom_C | trevarthan: you're honestly telling me your endpoint cant support a single g.711 call? |
| 00:17.21 | trevarthan | I can't remember exactly how it worked out, but I remember from testing (over a year ago) that the bitrate of the codec was basically half (or was it 1/3rd) the effective bandwidth used in reality. |
| 00:17.57 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
| 00:18.09 | [1]qrao | here are some stats on real-word bandwidth usage for diff codecs: http://www.cisco.com/warp/public/788/pkt-voice-general/bwidth_consume.html |
| 00:18.38 | [1]qrao | i meant g.723 is a good lbc not 726 |
| 00:18.49 | trevarthan | I remember liking iLBC because it didn't clog up my DSL up-stream bandwidth. I could still do other things on the wire without worrying about voice quality. |
| 00:19.54 | [1]qrao | g.729 will give u nearly the same utilization |
| 00:20.24 | [1]qrao | g.723 is even better tho |
| 00:24.25 | trevarthan | yeah, so g.729 and g.723 need a license though, right? |
| 00:25.13 | *** join/#asterisk bkw__ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 00:28.26 | trevarthan | wow, it's pretty difficult figuring out who to pay for those codecs.... |
| 00:29.00 | trevarthan | ah, found it: http://www.dspg.com/technology/LicensePricing.html |
| 00:29.02 | glm2k | digium for the former... |
| 00:30.17 | trevarthan | so I would have to pay 30k just to legally put g.723 on my server? |
| 00:32.09 | *** join/#asterisk JakBeatZ (n=JakBeatZ@beta.arionetworks.ca) |
| 00:35.19 | JakBeatZ | Anyone here have a 7970 working with SIP? Having an issue with getting a SIP load on it. It seems to be running SCCP 5.0.3S. When I have a SEP<MAC>.cnf.xml file in place, the phone request the file, the TFTP server transmits it, but even though it has the loadInformation line in the SEP file, the phone doesn't upgrade -- all I get is Load ID Incorrect. If I remove the SEP file, the phone requests XmlDefault.cnf.xml. With the loadInforma |
| 00:35.20 | JakBeatZ | tion line in that file, the phone requests the Xml file, but I get no errors at all; no Load ID Incorrect error and no upgrade happening. Anyone have any ideas? |
| 00:36.40 | *** join/#asterisk JakBeatZ (n=JakBeatZ@beta.arionetworks.ca) |
| 00:36.48 | JakBeatZ | Whoops :/ |
| 00:37.02 | trevarthan | ok, so the digium g.729 prices are pretty reasonable. Is that yearly? Or one-time? |
| 00:37.38 | trevarthan | And I assume I would be able to use those g.729 licenses on sip channels as well as zap? |
| 00:39.46 | mog | one time |
| 00:39.54 | mog | you cant use g729 on zap |
| 00:40.08 | mog | ulaw /alaw only on tdm |
| 00:41.42 | trevarthan | mog: yeah, that makes sense. The buy page says it works with zap cards which confused me. :) |
| 00:42.13 | mog | people often use zap to iax/sip/h323 etc |
| 00:42.26 | mog | codec does g729 to slin |
| 00:42.27 | trevarthan | mog: yeah, I get it. |
| 00:42.33 | mog | so it can do anything |
| 00:42.40 | mog | thats why it says that |
| 00:42.47 | mog | but it doesnt matter technology |
| 00:43.19 | symlink | mog: orange! |
| 00:43.30 | mog | symlink, apple? |
| 00:43.46 | symlink | hrm, my Digium shirt does not look apple colored :( |
| 00:48.44 | *** join/#asterisk linagee (n=na@cpe-66-75-142-207.san.res.rr.com) |
| 00:52.05 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 00:59.07 | *** join/#asterisk kph100 (n=kph100@206-248-156-72.dsl.teksavvy.com) |
| 00:59.10 | kph100 | hello |
| 00:59.41 | kph100 | is there way to dynamically create contexts in asterisk realtime extensions? |
| 01:04.17 | *** join/#asterisk arcanine (n=arcanine@203.82.44.179) |
| 01:04.46 | arcanine | hi |
| 01:04.57 | arcanine | can anyone help me wit my asterisk |
| 01:07.37 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:07.57 | arcanine | hi kumbang |
| 01:08.06 | Druken | arcanine: think you can be a little more specific? |
| 01:08.52 | *** join/#asterisk techie (n=techie@ppp-69-239-205-253.dsl.frs2ca.pacbell.net) |
| 01:09.50 | zimdog | Is there a way to set the messages/voicemail button on an IP500 to dial *97 and go directly to voicmailmain |
| 01:09.55 | arcanine | i wanted to use dialogic d600-jct2e1 and dialogic vfx card |
| 01:09.59 | arcanine | is it possible |
| 01:11.06 | arcanine | to use them in ivrs and do call outbound |
| 01:11.21 | pfn | it's scary that people are using asterisk to phish now... |
| 01:11.37 | pfn | asterisk/voip |
| 01:17.31 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 01:18.28 | *** join/#asterisk evildeshi (n=evildesh@pool-71-98-10-135.mdsnwi.dsl-w.verizon.net) |
| 01:19.43 | evildeshi | can someone help me out with a compile error im getting with asterisk on debian sarge with kernel 2.4? http://pastecode.com/4543 is the error im getting |
| 01:26.03 | Druken | pfn: phish ? |
| 01:26.32 | *** join/#asterisk type0 (i=type0@148-220-223-66.gci.net) |
| 01:26.50 | type0 | anyone know of a good gui tool for asterisk? *HOPEFULLY FREE?* |
| 01:27.08 | pfn | yeah, instead of giving bogus email addresses, they're using bogus 800 phone numbers and recording the conversations in a mitm attack |
| 01:27.24 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:27.25 | trevarthan | type0: trixbox is good. |
| 01:27.48 | Druken | pfn: oh joy... loads of fun... |
| 01:28.00 | type0 | will it run on anything with glibc or what? |
| 01:28.12 | bkw__ | LTNS boi |
| 01:28.14 | bkw__ | where you been? |
| 01:28.52 | type0 | also.. will trixbox use zaptel? |
| 01:28.55 | trevarthan | type0: it's a complete OS on a CDROM. |
| 01:29.06 | type0 | ahhhh |
| 01:29.13 | trevarthan | type0: yes, it uses ZAP. |
| 01:29.14 | type0 | i need something I can run on my centos colo |
| 01:29.22 | pfn | hey |
| 01:29.34 | *** join/#asterisk annonimous (n=annonimo@189.136.62.17) |
| 01:29.48 | trevarthan | type0: how about freepbx? That's one of the many GUI packages included with trixbox. Probably the most important one. |
| 01:30.09 | type0 | yeah but if trixbox is an OS |
| 01:30.09 | trevarthan | type0: BTW, trixbox is centos based. |
| 01:30.14 | type0 | hmm |
| 01:30.14 | type0 | well |
| 01:30.17 | type0 | I dont have access to the colo |
| 01:30.20 | type0 | im in alaska.. its in california |
| 01:30.35 | type0 | ive got a teliax account setup |
| 01:30.39 | trevarthan | type0: freepbx is just a web GUI. not an OS. |
| 01:30.42 | type0 | ok |
| 01:30.50 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 01:30.53 | type0 | so freepbx is a gui for asterisk? |
| 01:31.06 | bkw__ | asterisk is already gui |
| 01:31.10 | trevarthan | type0: more specifically, it's a configuration GUI. |
| 01:31.24 | Qwell | You want a gui, use xemacs |
| 01:32.47 | mog | lol |
| 01:33.07 | Qwell | mog! |
| 01:33.09 | *** join/#asterisk `Tingles` (n=tingles@S01060011d8ecb1d0.cg.shawcable.net) |
| 01:33.39 | zimdog | Is there a way to set the messages/voicemail button on an IP500 to go directly to voicemail? |
| 01:34.25 | mog | gtk emacs is better Qwell |
| 01:34.34 | Qwell | mog: xterm + nano |
| 01:34.35 | bkw__ | oh shut up |
| 01:34.37 | bkw__ | nano sucks |
| 01:34.43 | bkw__ | all editors suck |
| 01:34.54 | bkw__ | we need one editor to rule them all |
| 01:34.56 | mog | emacs and vim rock |
| 01:34.57 | Qwell | ed |
| 01:34.59 | bkw__ | wait a tick |
| 01:35.01 | mitcheloc | visual studio? |
| 01:35.02 | mog | no ed |
| 01:35.03 | Qwell | bkw__: ed is the standard editor |
| 01:35.09 | bkw__ | "the" standard? |
| 01:35.12 | Qwell | yes |
| 01:35.16 | bkw__ | in what book did you get that misconception out of? |
| 01:35.19 | annonimous | i have a trunk with callerid supported by provider but, the softphone doesnt show the caller number any idea how to fix it? |
| 01:35.23 | mog | man ed |
| 01:35.29 | Qwell | google that phrase LD |
| 01:35.30 | bkw__ | man sex |
| 01:35.30 | Qwell | :D |
| 01:35.41 | Qwell | http://www.gnu.org/fun/jokes/ed.msg.html |
| 01:35.44 | Qwell | pretty damn funny |
| 01:39.20 | bkw__ | http://bugs.digium.com/view.php?id=5162 |
| 01:39.24 | bkw__ | thats a hoot |
| 01:39.56 | trevarthan | VIM rules them all. grow some balls and use a real editor. |
| 01:40.09 | bkw__ | vim is ass |
| 01:40.21 | annonimous | vim-enhanced its good =D! |
| 01:40.46 | bkw__ | emacs in vim mode now thats funny |
| 01:42.30 | trevarthan | I'd use Visual Studio if it had a VIM emulation mode, but it's too damn slow without it. |
| 01:45.29 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 01:48.37 | mog | vim rocks man |
| 01:48.46 | mog | vim and emacs just plain fun |
| 01:51.14 | ido | nvi |
| 01:54.14 | *** join/#asterisk LoneShadow (n=duh@59.92.152.69) |
| 01:55.36 | Strom_C | mog: I should take a photo of this thai restaurant near my apartment called "Vim" |
| 01:55.50 | Qwell | Strom_C: nice |
| 01:56.52 | *** part/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 01:57.03 | mog | ooh /me wants it |
| 01:58.31 | *** join/#asterisk tengulre (n=tengulre@222.90.66.156) |
| 02:01.09 | Strom_C | i'll take it tomorrow when it's not dark out |
| 02:05.47 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 02:06.59 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 02:07.53 | niter3 | Strom_C: I've connected to somebody's asterisk system from my own with register => user:password@domain and added the username in sip.conf and an extension that routes through the remote asterisk box. However, the user can't call me... I get an authentication fail. Will I have to add a user on my end in my sip.conf that allows him to call me now? |
| 02:08.08 | niter3 | He's simply just trying to call the extension I connected to and it bitches.... |
| 02:09.46 | Strom_C | niter3: I don't recall our previous conversations? |
| 02:09.58 | niter3 | I was speaking with you yesterday sorry. |
| 02:10.09 | niter3 | Figured i ask. |
| 02:11.09 | Strom_C | please jog my memory of what we wre trying to debug |
| 02:11.25 | niter3 | freeworld dialup |
| 02:11.45 | Strom_C | oh |
| 02:11.49 | Strom_C | i dont recall doing much |
| 02:11.56 | Strom_C | just asking you to try type=friend |
| 02:12.00 | Strom_C | and then you disappeared |
| 02:12.05 | niter3 | yep that worked. |
| 02:12.05 | Strom_C | or type=peer or something |
| 02:14.52 | *** join/#asterisk janeNarak (n=chonlada@jane.lru.ac.th) |
| 02:17.21 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 02:18.05 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:18.10 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:19.14 | TheCops | Hi, Someone is using headset with Polycom IP601 ? i have some problem with the headset, people hear me so far away. |
| 02:19.17 | niter3 | you wouldn't have an idea would you? |
| 02:19.52 | [TK]D-Fender | Hey, trying to compile the latest Zaptel and getting an error concerning the VPM for which I think I heard something about vreifly. Can someone point out the quick fix if any for this? |
| 02:20.43 | symlink | [TK]D-Fender: grab a checkout from SVN, the tarballs need to be rebuilt |
| 02:21.15 | [TK]D-Fender | symlink : ! ! ! |
| 02:21.26 | symlink | [TK]D-Fender: that'll be $50! |
| 02:21.31 | niter3 | Anyone have a clue how I woul do this? |
| 02:21.37 | niter3 | Would I have to make another extension? |
| 02:21.38 | [TK]D-Fender | OMGZ. |
| 02:21.46 | niter3 | for himt o register to? |
| 02:21.50 | bkw_ | grrr |
| 02:23.18 | *** join/#asterisk dools (n=iain@socata.scoastnet.com.au) |
| 02:24.09 | [TK]D-Fender | Dang, now this box is missing subversion... whats the package to grab for Centos 4.3? |
| 02:24.12 | dools | i've run asterisk using safe_asterisk. the port in manager.conf is 5038, but when i try to do telnet 127.0.0.1 5038 i get connection refused. does anyone know why asterisk might be refusing this connection or some way of testing out connectivity on that port? |
| 02:24.12 | Nugget | telnet is eeeeeeevil! |
| 02:24.57 | JT | err |
| 02:25.26 | JT | he's obviously using telnet as a testing tool |
| 02:25.31 | JT | which is perfectly fine |
| 02:25.34 | JT | not as a remote shell |
| 02:25.35 | [TK]D-Fender | Ahh.. found it |
| 02:25.52 | [TK]D-Fender | JT.... you're talking to the Nugget-bot ;) |
| 02:26.10 | dools | yes i'm actually connecting using php's fsockopen, but it's not working so i'm trying to work out why |
| 02:26.18 | JT | heh |
| 02:28.11 | tengulre | dools: are you sure [general] enable=yes in manager.conf file? |
| 02:28.53 | tengulre | dools: using netstat-na | grep 5038, |
| 02:29.49 | dools | tengulre: it was set to no. thanks |
| 02:30.58 | tengulre | dools: :) |
| 02:31.39 | `Tingles` | anyone have any issues with voicemail not accepting passwords? even thoe i have verified it in the conf files.. and it used to be working until one user changed a password ... now all passwords for all mailboxes are invalid |
| 02:31.51 | dools | tengulre: someone else actually set this up, i'm just installing some sms features on top of it |
| 02:34.45 | drfreeze | hi |
| 02:35.28 | drfreeze | anyone with Polycom 301 provisioning experience around? |
| 02:35.48 | Strom_C | yep |
| 02:35.59 | [TK]D-Fender | symlink : Any idea when they are going to patch the tarball? |
| 02:36.03 | *** part/#asterisk Here_And_There (n=Here_And@pool-71-246-0-110.phlapa.fios.verizon.net) |
| 02:36.05 | [TK]D-Fender | drfreeze : Just ask |
| 02:36.08 | bkw_ | patch? |
| 02:36.10 | bkw_ | hatch? |
| 02:36.16 | bkw_ | chicken + egg? which came first |
| 02:36.22 | drfreeze | I can't get my phones to register |
| 02:36.34 | [TK]D-Fender | bkw_ : The chicken is the egg so the answer is YES |
| 02:36.43 | drfreeze | I have several 501's that I have deployed. The work fine |
| 02:37.03 | drfreeze | I'm thinking it is something in the phones menu that I am not doing right |
| 02:37.09 | bkw_ | So who puts the Risk in "Asterisk"? |
| 02:37.19 | drfreeze | bkw_: aster |
| 02:37.26 | bkw_ | hehe |
| 02:37.26 | [TK]D-Fender | bkw_ : Milton Bradley of course! ;0 |
| 02:37.28 | bkw_ | good answer |
| 02:37.36 | bkw_ | [TK]D-Fender: even better answer |
| 02:37.37 | Strom_C | do the phones pick up the correct settings and just fail to register, or do they not pick up the new settings at all? |
| 02:37.56 | type0 | ok.. so no one is answering me in the freepbx channel.. Fatal error: Call to undefined function: _() in /home/glacial/public_html/admin/index.php on line 18 |
| 02:38.03 | type0 | is that a PEAR problem? |
| 02:38.15 | drfreeze | Strom_C: the phones get an IP and are listed as unregistered |
| 02:38.26 | bkw_ | type0: I suspect its a PHP related issue |
| 02:38.30 | Strom_C | but do the line appearances pick up the correct numbers? |
| 02:38.33 | [TK]D-Fender | type0 : And you'll gett even LESS of an answer here as the channel topic suggests... |
| 02:38.35 | dools | i know the answer to the chickend and the egg issue |
| 02:38.46 | bkw_ | BOSTON HERE I COME... at 7:11 am tomorrow |
| 02:38.56 | orlock | dools: so do i. |
| 02:38.59 | orlock | the egg came first |
| 02:39.03 | dools | yep |
| 02:39.11 | orlock | reptiles were hatching from egs way before birds were |
| 02:39.21 | dools | haha no! |
| 02:39.25 | dools | that's not it |
| 02:39.32 | drfreeze | Strom_C: ahhh. No. Are they supposed to? |
| 02:39.46 | Strom_C | drfreeze: well there would be part of the proble, |
| 02:39.47 | dools | type0: that looks like there is a variable function call but the variables are missing, what's on that line of code? |
| 02:39.48 | Strom_C | er, problem |
| 02:40.00 | Strom_C | drfreeze: are you provisioning them via ftp or tftp? |
| 02:40.29 | drfreeze | manually. I only have 3 301's |
| 02:40.48 | dools | orlock: the chicken egg comes before the chicken, because each chicken that is born is ever so slightly evolutionarily different that the one that hatched the egg. so for any given evolutionary version of a given chicken, the chicken egg preceded it |
| 02:40.55 | Strom_C | .....manually? |
| 02:41.24 | drfreeze | Strom_C: through the advanced menu on the phone |
| 02:41.27 | drfreeze | Strom_C: Addr->IP : (Unspecified) Port 0 |
| 02:41.33 | Strom_C | drfreeze: are you bonkers? |
| 02:41.38 | Strom_C | drfreeze: use tftp or something |
| 02:41.47 | [TK]D-Fender | Strom_C :chan_redundant |
| 02:41.52 | Strom_C | heh |
| 02:42.23 | drfreeze | Strom_C: why. where do I get the file for tftp? There were only about 5 things to define for the 501's |
| 02:42.45 | Strom_C | [TK]D-Fender: you've got that link hander than I do |
| 02:42.54 | Strom_C | freedomphones.net/something/somethingelse |
| 02:43.06 | Strom_C | s/hander/handier/ |
| 02:43.24 | Strom_C | http://www.freedomphones.net/polycom/files/ |
| 02:43.25 | Strom_C | there we go |
| 02:44.36 | *** join/#asterisk MikeJ (n=_MikeJ@d14-69-8-30.try.wideopenwest.com) |
| 02:45.31 | [TK]D-Fender | Strom_C : I look for it a little every time :) I host my own so I tend to forget other peoples stuff.... |
| 02:45.59 | [TK]D-Fender | I tend to have it first because I dog the shit outta my supplier :) |
| 02:46.07 | Strom_C | i forgot I had it in my bookmarks |
| 02:51.01 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.152.112) |
| 02:52.21 | *** join/#asterisk mrgoby (n=mrgoby@c-68-42-71-60.hsd1.mi.comcast.net) |
| 02:53.15 | *** join/#asterisk compu73rg33k (n=compu73r@d53-64-54-188.nap.wideopenwest.com) |
| 02:53.21 | compu73rg33k | anybody here have an asterisk running on ubuntu? |
| 02:53.52 | Qwell | compu73rg33k: What's your real question? |
| 02:54.27 | [TK]D-Fender | Qwell : Who killed J.R.? |
| 02:54.33 | Qwell | [TK]D-Fender: I did |
| 02:54.35 | Qwell | shh |
| 02:54.49 | mrgoby | howdsies... looking on the wiki, settting up call transfer on a little pbx... is GOTO_ON_TRANSFER the var to set ? |
| 02:55.06 | Qwell | mrgoby: huh? |
| 02:55.39 | compu73rg33k | Qwell: that's is my question lol. I was wonderering if anybody was running * on ubuntu and if they are, whether it was ubuntu desktop or server |
| 02:55.45 | mrgoby | i'm having issues... just hangs up on successful transfer for the transferer... i want it to dump the transferer to a specific context/exten/priority |
| 02:56.20 | Qwell | compu73rg33k: Linux is Linux |
| 02:56.23 | compu73rg33k | Qwell: trying to decide whethr to install ubuntu server, ubuntu desktop, or slackware on my new server that will be a LAMP, *, and light desktop used computer |
| 02:56.45 | mrgoby | compu73rg33k: you prolly shouldnt run asterisk on a desktop (or with X) in general... |
| 02:57.04 | compu73rg33k | hmmm why is that mrgoby |
| 02:57.30 | mrgoby | at least that has been my experience... works good for testing, etc... but X and * will fight over interupts |
| 02:57.34 | bkw_ | hey Qwell |
| 02:57.38 | bkw_ | I talked to you know who |
| 02:57.40 | hacked`` | guys, how come when i go to freepbx, add a trunk, then save it, and then i go and do a sip show registry, it doesnt show the trunk i added |
| 02:58.00 | [TK]D-Fender | compu73rg33k : You can get * running an just about any sane linux setup. Ubuntu is by default lacking MANY of the required packages to compile * so its a bumpier road |
| 02:58.26 | compu73rg33k | ah interesting [TK]D-Fender what distro are you using? |
| 02:58.33 | [TK]D-Fender | compu73rg33k : But if you know what you're doing you can get the job done. Its just more work and hassle. Slackware = pretty much instant success. |
| 02:58.54 | [TK]D-Fender | compu73rg33k : I run it on slackware personally and CentOS for most of my clients. |
| 02:59.43 | compu73rg33k | Hmm now I'm thinkign whether or not to put * on a separate computer than my file server heh *sigh* the many options |
| 03:00.02 | compu73rg33k | becuase ubuntu server seems very quick to setup a lamp server |
| 03:00.54 | mrgoby | so, how do i set the context and exten for the transferer to land on using a blind (#) transfer ? |
| 03:01.03 | [TK]D-Fender | compu73rg33k : Slackware w/ "everything" = instant LAMP server as well. Do what you feel most comforatable with, |
| 03:01.10 | mrgoby | it will only hangup on me right now |
| 03:01.30 | [TK]D-Fender | mrgoby : Its the same context as every other call placed fromt hat device. |
| 03:01.50 | mrgoby | how do you set the extension then ? |
| 03:02.06 | [TK]D-Fender | mrgoby : Set what extensions? |
| 03:02.07 | mrgoby | is it +101 or +102 or ? |
| 03:02.29 | [TK]D-Fender | mrgoby : You are completely "off the page" in your understanding of transfers. |
| 03:03.04 | [TK]D-Fender | mrgoby : The phone doing the tranfering can call anywhere its allowed to normally and will then re-attach the call in progress to the new call to pass it off. |
| 03:03.19 | [TK]D-Fender | mrgoby : Effectively speaking its the same as a new call. |
| 03:03.52 | ziwapandey1980 | whois mrgoby |
| 03:04.07 | mrgoby | i am |
| 03:04.20 | mrgoby | who are you ziwa ? |
| 03:04.31 | ziwapandey1980 | its me ajit |
| 03:04.46 | ziwapandey1980 | i m looking for kram? |
| 03:05.44 | mrgoby | so, TK: scenario: i am in context 1, which has context 2 included in it. context 2 has a pattern match which triggers a dial... the person i dial, i transfer... do i land in the context2 exten + 1 ? |
| 03:05.59 | mrgoby | priority + 1 i mean |
| 03:06.31 | mrgoby | ~jbot seen kram |
| 03:06.49 | jbot | kram <n=mark@pdpc/sponsor/digium/kram> was last seen on IRC in channel #asterisk, 56d 11h 13m 13s ago, saying: '*cries*'. |
| 03:06.49 | mrgoby | where is jbot ? |
| 03:06.50 | mrgoby | jbot: dude |
| 03:07.41 | bkw_ | wonder why he was crying |
| 03:07.44 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 03:08.01 | mrgoby | bkw_: check the changelogs from 56 days ago :) |
| 03:09.01 | [TK]D-Fender | mrgoby : When you dfinish your transfer your end of the call gets DISCONNECTED |
| 03:09.16 | ziwapandey1980 | . |
| 03:09.35 | mrgoby | ok, so you can't transfer and then 'fork' the call ? |
| 03:09.47 | bkw_ | mrgoby: tell me |
| 03:10.31 | mrgoby | bkw_: naw, i dunno... just guessing that would be the reason |
| 03:11.12 | mrgoby | so, TK: you are saying that the transferee is basically taking over the call channel and that the transferer always gets hungup on ? |
| 03:11.41 | `Tingles` | anyone have any issues with voicemail not accepting passwords? even thoe i have verified it in the conf files.. and it used to be working until one user changed a password ... now all passwords for all mailboxes are invalid |
| 03:11.44 | `Tingles` | any ideas? |
| 03:12.44 | mrgoby | Tingles: i think asterisk edits the actual config file when they change the password, so i would imagine that would be the place to check... if not, did you try restarting asterisk ? |
| 03:12.54 | `Tingles` | tried both |
| 03:13.01 | `Tingles` | password in the conf file has not changed.. |
| 03:13.06 | `Tingles` | and restarted asterisk |
| 03:13.14 | `Tingles` | its kinda strange.. cause all the sudden no one can login |
| 03:13.17 | `Tingles` | not jsut that user.. |
| 03:13.22 | mrgoby | perms ? |
| 03:13.30 | mrgoby | wild guess |
| 03:13.35 | `Tingles` | hasn`t changed... |
| 03:15.10 | `Tingles` | ya its kinda strange... |
| 03:15.16 | `Tingles` | any idea how i can reset a users password? |
| 03:15.26 | `Tingles` | either then not changing the conf file as it doesn`t do anything.. |
| 03:16.21 | [TK]D-Fender | mrgoby : thats the idea. When you transfer a call you are getting RID of it. |
| 03:19.18 | *** part/#asterisk dools (n=iain@socata.scoastnet.com.au) |
| 03:21.59 | Toadyus | does anyone know how u become able to por local #'s? |
| 03:22.06 | Toadyus | do you have to buy a pri? |
| 03:23.14 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-166-30.red.bezeqint.net) |
| 03:23.55 | *** join/#asterisk Frogdude (n=chris@c-24-16-72-159.hsd1.mn.comcast.net) |
| 03:25.54 | ziwapandey1980 | is there any one using vicidial ? |
| 03:26.18 | raidenz | Does anyone know how to get the CDR(duration) in an 'h' extension using latest asterisk 1.2.12? It keeps saying that the duration of the call is 0. ( ${CDR(duration)} ) |
| 03:34.26 | docelmo | ask in #dev.. they probably broke it. |
| 03:34.52 | raidenz | good idea. thanks |
| 03:35.42 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 04:03.29 | *** join/#asterisk HaMYaI (i=HaMYaI@202.8.86.164) |
| 04:06.08 | arcanine | have anyone used centos on asterisk? |
| 04:07.18 | HaMYaI | arcanine, #FreePBX |
| 04:07.34 | HaMYaI | arcanine, asterisk on CentOS |
| 04:07.46 | arcanine | yah |
| 04:08.28 | JT | HaMYaI: i thought arcanine asked about asterisk on centos, not freepbx |
| 04:08.54 | HaMYaI | JT, you're wrong probably |
| 04:09.06 | JT | HaMYaI: why's that? |
| 04:09.45 | HaMYaI | JT, I bet he's using FreePBX and is having a problem with it |
| 04:10.10 | arcanine | using asterisk on centos |
| 04:10.11 | JT | what makes you bet that? |
| 04:12.06 | HaMYaI | JT, well I've seen most people using asterisk on CentOS are using FreePBX |
| 04:12.11 | HaMYaI | JT, that's all |
| 04:12.27 | JT | ok, well it seems he's not |
| 04:12.33 | JT | you know it's easier to just ask :) |
| 04:12.34 | HaMYaI | JT, I mean TrixBOX |
| 04:12.56 | HaMYaI | arcanine, are you using trixbox? |
| 04:13.07 | HaMYaI | =) |
| 04:14.01 | arcanine | nope |
| 04:15.09 | HaMYaI | arcanine, well I thought I could help |
| 04:15.13 | HaMYaI | =( |
| 04:15.16 | Dovid | morning |
| 04:15.21 | Dovid | whats the problem |
| 04:18.34 | fearthepenguin | . |
| 04:19.39 | [TK]D-Fender | arcanine : Don't know about using CentOS on Asterisk, but i do know plenty about running Asterisk on CentOS ;) |
| 04:19.53 | Dovid | i luv the sent os ;) |
| 04:21.05 | ajitekno | hi Dovid |
| 04:21.05 | [TK]D-Fender | CentOS works just great in my experience. I suggest it to my clients and the install works great |
| 04:21.22 | ajitekno | i wish to use pound key, it is good idea ? |
| 04:21.26 | Dovid | ajitekno: hi |
| 04:21.35 | Dovid | # for what ? |
| 04:21.40 | Dovid | _action test |
| 04:21.46 | ajitekno | asterisk with vicidial |
| 04:21.59 | Dovid | for what function ? an extension ? |
| 04:22.29 | ajitekno | h |
| 04:24.06 | Dovid | how would u use the # for the h exten ? |
| 04:24.25 | orlock | Hmm.. i am going to probably get my head bitten off for asking.. |
| 04:24.32 | orlock | but is anybody here using asterisk rpm's? |
| 04:24.57 | ajitekno | i say using # for vicidial |
| 04:25.13 | ajitekno | predictive dialer |
| 04:25.54 | *** join/#asterisk matt_ (n=Matt@82-33-68-44.cable.ubr01.trow.blueyonder.co.uk) |
| 04:26.31 | ajitekno | hi Matt |
| 04:26.43 | matt_ | hello, does anybody know of any equipment that is SIP or IAX compatiable and will allow me to put a custom sim card into a mobile and have it connect to this device |
| 04:26.51 | matt_ | which will then link back to asterisk |
| 04:26.55 | matt_ | ajitekno: hi |
| 04:27.11 | matt_ | so you can like use cell phones as voip phones |
| 04:27.15 | ajitekno | plz go to 2n website |
| 04:27.23 | matt_ | huh ? |
| 04:28.05 | ajitekno | matt_, i wish to use pound key linux from rpath with vicidial |
| 04:28.13 | ajitekno | is it good idea? |
| 04:28.49 | matt_ | i dont know |
| 04:29.03 | matt_ | i dont know what vicidial is |
| 04:29.10 | matt_ | a softphone i guess ? |
| 04:29.28 | ajitekno | sorry i thout u as matt flowrel |
| 04:30.04 | ajitekno | mya be u can end yor search here |
| 04:30.05 | ajitekno | http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html |
| 04:31.32 | matt_ | i have seen that and devices like it, they allow you to dial out to the gsm networks and get calls from the gsm networks |
| 04:31.38 | matt_ | i wanna like run my own mini gsm network |
| 04:31.58 | matt_ | so i write the sims, put them in the phones and the phones connect back to this box |
| 04:32.08 | matt_ | which is linked to asterisk |
| 04:40.33 | *** join/#asterisk Frogdude (i=Frogdude@c-24-16-72-159.hsd1.wa.comcast.net) |
| 04:44.00 | *** join/#asterisk LoneShadow (n=duh@59.92.132.114) |
| 04:47.13 | stephane_ | jour |
| 04:49.17 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 04:49.40 | [TK]D-Fender | Hey, new little problem. Chan_zap doesnt' seem to compile with *. I used SVN for zaptel beacause of that bug, but FTP for * 1.2.12. Is that why it doesn't pick it up? |
| 04:52.12 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 04:52.50 | *** join/#asterisk tlow (n=tlow@bgp.terrorist.net) |
| 05:00.54 | *** join/#asterisk DEader (i=Messiah@ool-44c6ac15.dyn.optonline.net) |
| 05:00.57 | DEader | hello |
| 05:01.10 | DEader | if i create a conference room on asterisk |
| 05:01.17 | DEader | and assign room number 7171 |
| 05:01.23 | DEader | how do i access the room |
| 05:03.34 | [TK]D-Fender | DEader : "show application meetme" |
| 05:03.35 | LoneShadow | dial 7171 |
| 05:03.50 | [TK]D-Fender | LoneShadow : Big assumptions...... careful.... |
| 05:04.04 | LoneShadow | why wouldnt dialing 7171 work ? |
| 05:04.32 | *** part/#asterisk tlow (n=tlow@bgp.terrorist.net) |
| 05:05.53 | [TK]D-Fender | LoneShadow : Who said he set up ANYTHING in extensions.conf? |
| 05:06.03 | [TK]D-Fender | Anyone got a hint for my zaptel problem? |
| 05:07.17 | LoneShadow | what is the meaning of assigning room # 7171 ? |
| 05:08.06 | LoneShadow | I have not created meetme in plain vanilla asterisk, only from freepbx, so dont have experience the usual way :D |
| 05:09.07 | [TK]D-Fender | LoneShadow : You might want to hold off on giving advise then :) |
| 05:10.27 | LoneShadow | [-_-] |
| 05:11.41 | *** join/#asterisk eject_ck (n=eject@195.95.232.148) |
| 05:12.07 | DEader | i dial 7171 but it didn't work |
| 05:13.11 | DEader | how do i give it extension |
| 05:13.34 | LoneShadow | ok I stand/sit corrected :D |
| 05:13.59 | LoneShadow | did you check voip-info.org, they usually have info |
| 05:15.24 | [TK]D-Fender | DEader : "show application meetme" <------ |
| 05:15.47 | DEader | ok |
| 05:16.50 | DEader | [general] |
| 05:16.50 | DEader | [rooms] |
| 05:16.50 | DEader | conf=>7171,12345,123456 |
| 05:16.58 | DEader | thats it |
| 05:17.23 | [TK]D-Fender | DEader : if you want to DIAL anything you should be working on extensions.conf now.... |
| 05:17.47 | hmmhesays | heh |
| 05:17.50 | hmmhesays | never fails |
| 05:18.14 | DEader | then how do i connect it to the room |
| 05:19.21 | tzafrir_laptop | [TK]D-Fender, which chan_zap? trunk requires zaptel trunk to build |
| 05:21.54 | *** join/#asterisk kusznir (n=kusznir@66-233-98-93.lew.clearwire-dns.net) |
| 05:22.05 | [TK]D-Fender | tzafrir_laptop : I'm using FTP * + Zaptel SVN (trunk?) |
| 05:22.25 | [TK]D-Fender | tzafrir_laptop : bad combo? |
| 05:26.16 | kusznir | Does anyone know an "external" port mapping web page I can use? My IAX broke, and I think it might be my privider "managing" (blocking) 4569 :( |
| 05:26.58 | *** join/#asterisk Rahail (n=dd@ip66-104-51-162.z51-104-66.customer.algx.net) |
| 05:27.24 | eject_ck | kusznir, simply change default port |
| 05:27.55 | kusznir | Will that work with privders whom are too large/unwilling to make any special config on their end? |
| 05:28.20 | eject_ck | ;) no |
| 05:28.53 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 05:28.54 | kusznir | didn't think so :( (I had the same problem with them blocking 5060 about 4 months ago.....) |
| 05:28.55 | eject_ck | tell abt your problem to u r ISP |
| 05:29.06 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 05:29.23 | kusznir | They're not much help....been there, done that (for other issues). I usually have to teach them how networks work, etc. |
| 05:29.30 | eject_ck | I had same problem |
| 05:29.36 | kusznir | err..how TCP/IP works |
| 05:30.48 | eject_ck | u can use tunnel to external router behind u provider network ? |
| 05:31.48 | kusznir | Possibly. However, if they've just changed their policies and blocked this port, I'd fight to get out of my contract and go somewhere else first :) |
| 05:32.41 | eject_ck | it's best way |
| 05:32.59 | eject_ck | write official letter them |
| 05:33.16 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.152.112) |
| 05:33.19 | eject_ck | kusznir, from where u are |
| 05:33.24 | kusznir | I actually wouldn't mind dumping them and switching providers, I'm just stuck in a contract right now, and without a really good reason to press to the local management, I'm stuck for anouther 1.4 years. |
| 05:33.57 | kusznir | eject_ck: ?? are you asking where I am located? |
| 05:42.54 | *** join/#asterisk phpboy (n=shane@196.26.21.106) |
| 05:43.49 | phpboy | hey all, how do I list my zap channels in zaptel.conf if I have more than 1 TDM400P card with 4FXO modules on? I'd imagine it would be 1-8 if I had 2 cards but that doesn't seem to be working |
| 05:43.50 | phpboy | :< |
| 05:44.06 | eject_ck | kusznir, yes |
| 05:44.37 | kusznir | I'm in washington state in the US (northwest part of the US) |
| 05:49.25 | tzafrir_laptop | [TK]D-Fender, I suppose so. Though I'm not sure exactly what caused that breakage |
| 05:49.47 | tzafrir_laptop | phpboy, 1-4 ? |
| 05:50.07 | tzafrir_laptop | phpboy, try xpp/utils/genzaptelconf |
| 05:59.13 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:04.06 | kusznir | Ok, I've verified that my provider is not blocking 4569 in any way I can test. |
| 06:04.46 | *** join/#asterisk mick_linux (n=mick@adsl-153-159-186.mia.bellsouth.net) |
| 06:06.42 | type0 | hey.. I just read that asterisk business edition comes with support for dialogic.. is that true? |
| 06:09.08 | DarKnesS_WolF | any one using java applet with astrisk |
| 06:09.09 | DarKnesS_WolF | ? |
| 06:13.34 | DarKnesS_WolF | type0: what is dialogic ? |
| 06:13.48 | _Vile | a card mfg |
| 06:15.13 | _Vile | to say the least |
| 06:17.30 | *** join/#asterisk VOIPoTD (n=td@h46093824.area4.spcsdns.net) |
| 06:17.55 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-232.range86-139.btcentralplus.com) |
| 06:19.31 | VOIPoTD | Any admims here looking for full-time work? Looking to add a more guys to team. |
| 06:20.04 | kusznir | Hi all: I'm having some strange problems with iax that I can't track down. Short version: outbound IAX sessions to 2 of my 3 trunks are accepted, then hang indefinately or time out. Incomming works fine. No config changes made (that I am aware of) to cause breakage (it used to mostly work -- see pastebin for details). |
| 06:20.07 | kusznir | http://pastebin.ca/166691 |
| 06:20.35 | kusznir | I'd greatly appreciate any help on this...I've put a few hours in, and have gotten absolutely nowhere...this definately seems like a very strange problem to me. |
| 06:23.03 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:23.14 | kusznir | My two trunks that are failing as above are voipjet and asterlink; fwd is the one that is returning immediately with busy. |
| 06:24.42 | DarKnesS_WolF | can i get the callerID on a zap channel and execute a part from the dial plan using system application without answering the call ? |
| 06:25.44 | kusznir | Config files worked completey fine with Asterisk 1.0.10 (IIRC --- the * included with OpenWRT); I upgraded to 1.2.7.1 and had some problems, restarted, and had all these problems (outbound calling is non-functional). |
| 06:25.59 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:26.07 | kusznir | DarKnesS_WolF: I think I saw a code snipit for that on voip-info.org |
| 06:28.00 | DarKnesS_WolF | kusznir: yes ur talking about this ? http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf ? |
| 06:28.41 | kusznir | Looks like it..I think you can extract parts of that and use it in other ways too. |
| 06:29.27 | DarKnesS_WolF | kusznir: i tried this but always i got ignoued i'm i mean i dosn't work |
| 06:29.33 | kusznir | I suspect if you make your s,1 line run the program, passing in ${CALLERIDNUM}, you should be able to accomplish what you want. You can optionally answer it later. |
| 06:31.24 | kusznir | Have you tried something as simple as running a system command that outputs the CID info to a text file to make sure you're getting the data you think? |
| 06:31.48 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:32.13 | kusznir | (I'm by no means an asterisk dialplan expert, so I don't claim to know the exact syntax to do this, but it seems like a logical next step. That and watching the * console with high verbosity for any dialplan errors. |
| 06:32.29 | kusznir | Any takers on my IAX2 trunk problem? http://pastebin.ca/166691 |
| 06:33.05 | DarKnesS_WolF | kusznir: http://pastebin.ca/166695 |
| 06:34.40 | kusznir | Looks like you're setting the CID info up front...that variable should be pre-populated on an incomming call. I'd kill line 1 to start with. |
| 06:35.01 | DarKnesS_WolF | kusznir: didn't understand what do u mean? |
| 06:36.29 | kusznir | Line one on your pastebin: you're setting the CID info to itself. This may be causing problems, and in any case appears to be wrong. The SetCID.. commands are to set CID info when you want to override the CID info already there. (any incomming call is supposed to have the CID variables pre-populated with something) |
| 06:38.40 | kusznir | I'd try http://pastebin.ca/166697 as debugging info (look at the temp file to see what got logged) |
| 06:39.09 | DarKnesS_WolF | kusznir: thx got what what u mean |
| 06:41.29 | *** join/#asterisk xnon (i=xnon@200.82.222.123) |
| 06:42.21 | DarKnesS_WolF | kusznir: it echo the correct number |
| 06:42.26 | DarKnesS_WolF | but it's stil didn't hangup |
| 06:42.33 | DarKnesS_WolF | it go on and answer the call normaly |
| 06:42.49 | evildeshi | anyone around that can help with an odd audio problem |
| 06:43.08 | evildeshi | im able to connect to meetme with a buddy on it as well i heasr the automated greeting fine but we cant hear each other |
| 06:44.16 | kmilitzer | ~pb |
| 06:44.23 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 06:44.24 | kmilitzer | ~pastebin |
| 06:44.25 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 06:44.51 | evildeshi | also when i dial up another extension to a client we cant hear each other there either any ideas where i should start im kinda new to this |
| 06:44.58 | kusznir | DarKnesS_WolF: try http://pastebin.ca/166700 |
| 06:45.14 | kusznir | oh, wait...I see the probl. |
| 06:45.17 | kusznir | hold on. |
| 06:45.32 | DarKnesS_WolF | kusznir: ok |
| 06:46.00 | kusznir | DarKnesS_WolF: try this: http://pastebin.ca/166702 |
| 06:46.16 | kusznir | (Pay close attention to the ordering of the ["${ on line 2. |
| 06:47.24 | kusznir | evildeshi: can you and your buddy call each other directly and hear each other? |
| 06:49.01 | kmilitzer | Has anyone of you seen something like that: http://pastebin.ca/166703? |
| 06:49.29 | kmilitzer | An error message like "BAD! BAD! BAD!" Looks strange to me ;) |
| 06:52.27 | *** join/#asterisk spliner (n=vpanayot@83.228.51.12) |
| 06:53.42 | tzafrir | kmilitzer, next time, please put a space after the link |
| 06:57.44 | *** join/#asterisk HaMYaI (i=HaMYaI@202.8.86.164) |
| 06:59.19 | HaMYaI | Zaptel 1.2.9 compiles successfully on my "2.6.12-2.3.legacy_FC3" but not "2.6.17-1.2174_FC5smp" |
| 06:59.35 | HaMYaI | anyone got it compiled on FC5? |
| 06:59.50 | HaMYaI | probably non-smp ones |
| 07:03.22 | *** join/#asterisk stagiaire (n=stagiair@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 07:04.20 | DarKnesS_WolF | kusznir: very strange error message u have |
| 07:07.47 | kusznir | DarKnesS_WolF: the problem is there really isn't an error message...It just starts to work, then hangs. No explanation. |
| 07:13.25 | kusznir | I was hoping one of the asterisk gurus like Qwell, twisted, file, etc. would jump in there and shed some light, at least where to start looking next...I don't even know what other tests to run or how to continue debugging this..... |
| 07:20.35 | docelmo | someone in tier 2 support trying to be funny |
| 07:22.07 | *** part/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 07:27.38 | Jeffjohnson | i have an quadband and an hfcpci card in my machine, no I get several "zaphfc: bchan rx fifo not enough bytes to receive! [...] probably a buffer overrun" messages, what is wrong? Is this a critical error? It works without problems |
| 07:27.40 | *** join/#asterisk linagee (n=na@cpe-66-75-142-207.san.res.rr.com) |
| 07:29.15 | tzafrir | Jeffjohnson, what version of bristuff? |
| 07:30.03 | Jeffjohnson | tzafrir: 0.3.0-pre-1s, version from trixbox |
| 07:30.04 | tzafrir | also: someone wrote on some debian list that a workaround was to change the signalling type from bri_cp_ptmp to bri_cpe |
| 07:30.29 | tzafrir | I figure noone really needs bri_cpe_ptmp |
| 07:30.40 | Jeffjohnson | tzafrir: i have signalling bri_cpe allready, cause i need it .) |
| 07:31.57 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:32.02 | Jeffjohnson | tzafrir: do you know it the error is critical? |
| 07:33.24 | tzafrir | Looks like it |
| 07:33.43 | tzafrir | hmm... it's the bchan.... |
| 07:33.46 | *** join/#asterisk Skymarshal (n=Skymarsc@pD9E121DE.dip0.t-ipconnect.de) |
| 07:33.55 | tzafrir | If it's critical you'll hear it in the audio, I guess |
| 07:34.54 | Skymarshal | Do I have to configure a context for a sip user (in sip.conf) or is this only necessary for friend and peer? |
| 07:35.53 | kusznir | A context is needed for user. It might use the context "default" if not specified, but * has to know where to dump the user when they call. |
| 07:36.43 | Skymarshal | kusznir: but a user can not be called, can he? I think that a user can only make calls. Wrong? |
| 07:38.10 | kusznir | Yes, that's exactly why they need a context. Where in the dial plan is his call request placed from? |
| 07:38.51 | Skymarshal | kusznir: Got you! I mixed it up. Thanks! |
| 07:38.52 | kusznir | For example, you can have a "public" phone that won't have access to long distance. How do you restrict that? You create a restricted context in the dial plan that does not include any long distance numbers, then you stick that phone in that context. |
| 07:39.36 | tzafrir | Skymarshal, a sip user needs a context. If you don't define one it will use the context "default". A sip friend is by definition also a sip user |
| 07:40.45 | tzafrir | (that is: if you don't define one, it will use the default context from the general section, or "default" if none is defined there as well) |
| 07:42.51 | kusznir | Ok, so my problem http://pastebin.ca/166691 does appear to be config-related somehow...I just took a working trixbox /etc/asterisk and dumped it on this (non-trixbox) system and it did more-or-less work (IAX worked fine; fixing trixbox-isms is the more-or-less part). |
| 07:42.56 | *** join/#asterisk anthonyl (n=anthonyl@c-67-167-214-149.hsd1.il.comcast.net) |
| 07:43.13 | kusznir | I'd greately appreciate it of someone can take a look and tell me what stuipid mistake I made and am not seeing.... |
| 07:43.45 | DarKnesS_WolF | kusznir: didn't work also |
| 07:43.52 | DarKnesS_WolF | it always over my number as a caller ID |
| 07:43.52 | DarKnesS_WolF | hm |
| 07:43.58 | DarKnesS_WolF | something wrong i'm doing |
| 07:45.06 | kusznir | Can you pastebin the lines you're using? |
| 07:47.01 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 07:50.22 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 07:52.38 | *** join/#asterisk kore (i=kore@mindwipe.org) |
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| 07:53.18 | *** part/#asterisk dflow (i=pch@h82-143-131-199-static.e-wro.net.pl) |
| 07:55.42 | *** part/#asterisk spliner (n=vpanayot@83.228.51.12) |
| 07:56.25 | phpboy | tzafrir: 1-4 works for one card without a problem... however that doesn't cover the other card |
| 07:57.14 | phpboy | so what I tried was 1-8 assuming it would recognize card 2 as 5-8 |
| 07:57.23 | phpboy | but this was not the case :/ |
| 07:59.47 | Jeffjohnson | tzafrir: hmm usb and zaphfc had shared interrupts... Ive deactivate the usb controller, not it seems to work without errors |
| 08:04.38 | Jeffjohnson | tzafrir: the error only had a small delay :E |
| 08:08.18 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 08:11.24 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:13.36 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 08:15.38 | phpboy | I'm happy that I got asterisk working successfully on FreeBSD |
| 08:15.43 | phpboy | with TDM400 cards |
| 08:15.44 | phpboy | :D |
| 08:17.00 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 08:21.50 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:35.23 | *** join/#asterisk postel (n=jp@Wikimedia/Postel) |
| 08:37.18 | *** join/#asterisk Geliman (n=scorpio@unaffiliated/drkshdw) |
| 08:43.37 | *** join/#asterisk maciq (i=macros@chw.casablanca.cz) |
| 08:44.05 | *** part/#asterisk my007ms (n=my007ms@217.139.224.194) |
| 08:47.38 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 08:54.00 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 08:57.20 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 09:00.50 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
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| 09:22.29 | *** join/#asterisk Cyt (n=danielcy@athedsl-111849.otenet.gr) |
| 09:24.28 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:24.42 | pif | hi, anyone deployed polycom 2.0.1 fw ? |
| 09:39.51 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:44.51 | HaMYaI | Zaptel 1.2.9 compiles successfully on my "2.6.12-2.3.legacy_FC3" but not "2.6.17-1.2174_FC5smp" |
| 09:44.53 | HaMYaI | anyone got it compiled on FC5? |
| 09:44.54 | *** join/#asterisk CtRiX (n=CtRiX@aretha.navynet.it) |
| 09:45.44 | *** join/#asterisk adorah (n=admin@84.94.122.156.cable.012.net.il) |
| 09:49.13 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 09:49.40 | *** join/#asterisk sergee (i=opera@195.94.224.197) |
| 09:50.36 | sergee | Mw3: Hello! you here? |
| 09:51.16 | *** join/#asterisk phearless (n=phear@host81-138-68-106.in-addr.btopenworld.com) |
| 09:51.45 | phearless | hello [*] folks ! |
| 09:54.37 | docelmo | ACK! |
| 10:00.21 | *** join/#asterisk ToxaP (i=ToxaP@194.187.128.88) |
| 10:00.33 | ToxaP | Hello |
| 10:01.12 | *** join/#asterisk oej (n=oej@12.38.214.2) |
| 10:03.10 | Mw3 | sergee: yeah |
| 10:04.32 | sergee | Mw3: can you execute this command on your mISDN box? grep MODULE_PARM /usr/src/linux/drivers/isdn/hardware/mISDN/hfc_multi.c |
| 10:05.24 | sergee | Mw3: probably you'll need to change "linux" - to something different, e.g.: "linux-2.6.16.9" or something like that |
| 10:09.12 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 10:17.01 | phearless | ARG ! |
| 10:17.27 | phearless | where can I set the asterisk server in a linksys/sipura SPA942 voip phone ? |
| 10:23.19 | phearless | anybody got any SPA-something phones ? |
| 10:34.46 | docelmo | phearless what are you looking for? |
| 10:35.15 | phearless | where can I set the hostname or IP of the server |
| 10:35.55 | phearless | I can set the user id, the passwd, the ext number, etc but not the asterisk server |
| 10:40.44 | phearless | ??????? |
| 10:48.52 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 10:49.11 | *** join/#asterisk CSerpent (i=daemon@62.49.253.91) |
| 10:49.16 | CSerpent | morning |
| 10:50.36 | phearless | got it |
| 10:50.39 | phearless | it is "proxy" |
| 10:53.17 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:59.08 | RoyK | ~new features |
| 10:59.09 | jbot | botAbusers<Loser *>.push_back(new features); |
| 10:59.17 | RoyK | :P |
| 10:59.57 | phearless | ~help |
| 11:00.11 | RoyK | ~lart phearless |
| 11:00.12 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 11:00.38 | CSerpent | :D |
| 11:06.44 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 11:10.15 | *** join/#asterisk SaTLaN32 (n=satlan32@212.150.142.211) |
| 11:10.18 | SaTLaN32 | hi |
| 11:10.31 | SaTLaN32 | i need help configuring asterisk to work with T.28 |
| 11:10.35 | SaTLaN32 | T.38 |
| 11:10.58 | CtRiX | termination or passthrough ? |
| 11:11.09 | SaTLaN32 | passthrou |
| 11:11.24 | SaTLaN32 | i have welteck GW and Mediant 2000 on the other side |
| 11:11.35 | CtRiX | so edit sip.conf and hope it works. |
| 11:11.54 | SaTLaN32 | the flow is: analog fax connected to welteck configured to work T.38 |
| 11:12.18 | SaTLaN32 | welteck is registered in the asterisk, and teh mediant is configured as a trunk |
| 11:12.32 | SaTLaN32 | the e1 from the mediant is conenected to my system (VM) |
| 11:12.43 | SaTLaN32 | what should i edit in the sip.conf? |
| 11:12.57 | SaTLaN32 | t38pt_udptl = yes ? |
| 11:13.02 | CtRiX | for example |
| 11:13.09 | SaTLaN32 | for each exten |
| 11:13.09 | CtRiX | that one |
| 11:13.09 | SaTLaN32 | ? |
| 11:13.24 | SaTLaN32 | i did but same thing.. |
| 11:17.40 | SaTLaN32 | any other idea? |
| 11:18.43 | RoyK | SaTLaN32: what version of *? |
| 11:19.40 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 11:19.51 | SaTLaN32 | 1.2.9.1 |
| 11:20.51 | RoyK | forget it |
| 11:20.59 | RoyK | t.38 passthrough is only in trunk |
| 11:21.03 | RoyK | and openpbx, that is |
| 11:22.35 | SaTLaN32 | i pass from an extension to a trunk |
| 11:22.45 | SaTLaN32 | what shoudl i configure in the trunk? |
| 11:22.50 | CtRiX | * trunk |
| 11:23.29 | SaTLaN32 | ? |
| 11:23.41 | benjk | openpbx does more than just passthrough |
| 11:24.52 | SaTLaN32 | mm i lost you guys.. |
| 11:26.44 | benjk | RoyK says that the version of * you are using doesnt support t38 passthrough yet |
| 11:27.01 | benjk | instead you need trunk |
| 11:27.04 | CtRiX | an probably won't |
| 11:27.19 | benjk | that's the version which the developers are adding stuff to all the time |
| 11:27.45 | SaTLaN32 | it's not the latest? |
| 11:27.54 | benjk | RoyK also says that you can use openpbx, which has t38 passthrough, and I added that it does full t38 not only passthrough |
| 11:28.09 | SaTLaN32 | Update Feb 2006: bug/patch 5090 now contains a T.38 pass-thru patch for Asterisk v1.2.4 and SIP. |
| 11:28.18 | benjk | it may be the latest release, but trunk is the developer "version" |
| 11:28.39 | benjk | its what will become a release some time in the future |
| 11:28.48 | stoffell | anyone ever had this issue: call to international cell phone gives chanunavail when not answered ? |
| 11:28.55 | benjk | or more precisely what will lead to a release at some point in some form |
| 11:29.24 | benjk | stoffell, you should always give all the info |
| 11:29.37 | RoyK | SaTLaN32: better try openpbx. it's far more mature there |
| 11:29.40 | benjk | like, what kind of techonolgy do you dial out on? |
| 11:29.52 | SaTLaN32 | what is it? |
| 11:29.53 | benjk | is it SIP? IAX? ISDN? POTS? |
| 11:29.55 | stoffell | benjk: sorry.. using pri (e1) |
| 11:30.15 | benjk | in that case pri intense debug will probably tell you why you get that response |
| 11:30.27 | benjk | asterisk console messages are liars |
| 11:30.34 | kmilitzer | Is there really no other way to reflect changes in the manager.conf then to completely restart asterisk? |
| 11:30.36 | benjk | never trust asterisk console messages |
| 11:30.39 | RoyK | prolly 'pri debug' will suffice |
| 11:30.44 | stoffell | benjk: yes, I have pri debug activated, and see the isdn messages, but now ... to debug them |
| 11:30.51 | RoyK | ~benjk |
| 11:30.54 | jbot | hmm... benjk is abusing the bot, or he is l33t with the bot, or but in reality benjk is teaching the bot, or he is truely trying to crash the bot |
| 11:30.59 | benjk | the only thing that gives you real feedback is the debug output |
| 11:31.39 | benjk | can you find the message you get back. the one which reflects the hangup? |
| 11:31.46 | benjk | the pri message I mean |
| 11:31.58 | stoffell | benjk: i have a sample output here http://pastebin.ca/164129 |
| 11:32.00 | RoyK | SaTLaN32: either use the trunk from svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk or use opbx |
| 11:32.53 | stoffell | benjk: on lin 238 the call starts |
| 11:33.03 | stoffell | and next thing is the hangup |
| 11:33.45 | benjk | can you do the same with pri intense debug |
| 11:33.55 | benjk | then pastebin the output again |
| 11:34.16 | stoffell | okay benjk , will do that.. brb |
| 11:34.26 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 11:36.38 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 11:36.38 | RoyK | ~opbx |
| 11:36.47 | RoyK | ~openpbx |
| 11:36.48 | jbot | i heard openpbx is an asterisk fork without asterisk's limitations of using other GPLed code. see http://openpbx.org/ for more info, or join #openpbx |
| 11:37.32 | CtRiX | Asterisk 1.4 is really just trying to catch up with OpenPBX.org |
| 11:38.32 | RoyK | someone should write an article about real comparison between opbx and *. it's kindof unclear |
| 11:40.13 | SaTLaN32 | question |
| 11:40.19 | SaTLaN32 | i want to install asterisk trunk |
| 11:40.22 | SaTLaN32 | i have this error |
| 11:40.33 | SaTLaN32 | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 11:40.41 | SaTLaN32 | when runing ./configure |
| 11:41.42 | Dr-Linux | what's new features in asterisk 1.4? |
| 11:41.49 | Vec | Anyone know if it is possible to label certain incoming channels (lines) with certain names, eg Sales, Support. which can be viewed from the asterisk flash operator panel ? |
| 11:42.32 | RoyK | Dr-Linux: a few new bugs |
| 11:42.33 | RoyK | := |
| 11:42.34 | RoyK | :) |
| 11:42.46 | Vec | So the operator can answer, "Hello this is Sales", or "Hello would you like support" |
| 11:42.48 | RoyK | Dr-Linux: well, there's the jitterbuffer for rtp, config parser rewrite |
| 11:43.24 | Dr-Linux | RoyK: jitterbuffer is fix or new feature? |
| 11:43.47 | Dr-Linux | i'm asking about any new feature, like any voicema/dialplan/queue ehnacement? |
| 11:44.41 | RoyK | it's a fixteaure |
| 11:44.52 | RoyK | that is, a feature that should have been there in the first place |
| 11:46.22 | benjk | how can they "rewrite" something they didn't have to begin with? |
| 11:49.46 | *** join/#asterisk dr_chaos (n=dr_chaos@83.240.187.13) |
| 11:52.17 | RoyK | benjk: heh. |
| 11:52.27 | RoyK | benjk: i paid for that jb thingie... |
| 11:52.41 | benjk | I meant the config parser |
| 11:52.49 | benjk | asterisk doesn't have a config parser |
| 11:53.11 | benjk | the only parser in there is the bison built parser for CLI commands |
| 11:53.19 | RoyK | hm. yes.. |
| 11:56.18 | Winkie | is there any reason i'm getting two Newchannel events for a single SIP channel? the first being State: Down and the second being State: Ringing ? |
| 11:56.25 | Winkie | I would have expected Newstatus |
| 11:56.35 | Winkie | uh, Newstate, even |
| 11:58.36 | Sonderblade | is there a way to customize how asterisk generates cdrs? |
| 11:59.14 | Winkie | Sonderblade: exactly what do you mean? |
| 11:59.42 | Sonderblade | Winkie: i need to add metadata for each call |
| 12:00.01 | Winkie | Sonderblade: ah, I'm not entirely sure. I gave up with CDRs and i'm writing my own call tracker |
| 12:00.36 | Sonderblade | Winkie: i guess that is the way |
| 12:01.07 | RoyK | show function CDR |
| 12:01.17 | RoyK | set(CDR(userfield)=asdf) |
| 12:01.29 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 12:01.50 | RoyK | the userfield is up to 256 chars iirc, so you'll have to pack stuff into there |
| 12:02.58 | Sonderblade | RoyK: sounds perfect thanks |
| 12:03.25 | Sonderblade | RoyK: exactly which CDR does that function set? |
| 12:03.47 | RoyK | erm. the current... |
| 12:03.50 | RoyK | of course |
| 12:04.18 | Sonderblade | but a call can span multiple CDRs |
| 12:05.00 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:05.08 | Druken | Sonderblade: well, if you setup your machine properly, it shouldn't.... |
| 12:05.57 | Sonderblade | Druken: it should to, each interaction with the pbx produces a new CDR |
| 12:06.36 | Druken | each DIAL would produce a new cdr... |
| 12:07.44 | Sonderblade | no, other events also produces cdrs |
| 12:09.46 | RoyK | for instance the "hangup" cdr |
| 12:13.26 | Druken | uhg... it's bloody cold in my office this morning.... |
| 12:14.45 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:16.09 | Godsey | is it possible to have meetme not prompt caller to say their name? |
| 12:16.15 | Cyt | Hi, please... I need to setup my asterisk for when I receive a call in a sip line X it rings my phone and says "you are receiving a call from line X" and then put the call thru! What's the dial plan application that I should use? |
| 12:16.43 | RoyK | Godsey: yes |
| 12:16.46 | RoyK | ~docs |
| 12:16.48 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:17.14 | Godsey | http://www.voip-info.org/wiki-Asterisk+cmd+Meetme |
| 12:17.23 | Godsey | is this the wrong place for me to find that information? |
| 12:17.59 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 12:18.44 | [TK]D-Fender | Godsey: "show application meetme" |
| 12:18.46 | *** join/#asterisk ToxaP (i=ToxaP@194.187.128.88) |
| 12:18.57 | Druken | anyone here that flies? i'm curious... if you fly within your country, is it any quicker gettin on and off the plane? |
| 12:18.57 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:19.09 | Godsey | oh thanks |
| 12:20.28 | Godsey | I guess I'm dense, or something else is asking for the callers name :) |
| 12:20.42 | Godsey | because I don't see an option to not prompt user to record their name. |
| 12:25.47 | Godsey | <PROTECTED> |
| 12:26.03 | Godsey | so I have to find how to disable introuser :) |
| 12:26.46 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 12:28.03 | Godsey | I don't have i or I defined |
| 12:29.36 | Druken | why would you want to disable that? i'd love to enable in trouser on my gf.... |
| 12:31.36 | benjk | Druken you got that wrong, you'd want to enable ex-trouser on your gf |
| 12:32.00 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 12:34.49 | *** join/#asterisk hank (n=hank@netwichtig.de) |
| 12:35.06 | hank | hi |
| 12:35.10 | *** join/#asterisk swytch (n=root@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 12:35.30 | FlatFoot | hello hank |
| 12:36.26 | swytch | hello. i wander if i can use a crossed E1 cable to "loop back" from a digium wilcard and back again .. |
| 12:36.27 | swytch | can a normal "crossed E1" cable be used? |
| 12:37.10 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 12:37.16 | Druken | benjk: hehe too true |
| 12:37.46 | RoyK | swytch: loopback is to loopback one E1 to itself. crossover is to connect two E1s |
| 12:37.53 | hank | im trying to setup an asterisk server for use in our company (im only testing atm though) using debian unstable and a billion isdn adapter using a hfcsusb chipset. at first id like to make sure the card is properly recognized and working. how do i do that? tried some things with capiinit but linux and isdn seems to be rather difficult to me. any advice? |
| 12:38.17 | benjk | hank, use BRIstuff-stable |
| 12:38.23 | benjk | will work out of the box |
| 12:38.25 | swytch | so can a normal "crossed E1" cable be used between two digium wildcard ports to provide connection, and then use the ports as TE and NT ?? |
| 12:40.00 | [TK]D-Fender | swytch: Yes, and its often done to test the port configs |
| 12:40.46 | swytch | [TK]D-Fender: nice. im going to do it to pass sip calls threw a wildcard and see if the octassic chip does any magic .. |
| 12:41.36 | hank | benjk: how do you know? |
| 12:41.47 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:41.54 | benjk | cause I am using the same card with BRIstuff-stable |
| 12:42.06 | swytch | [TK]D-Fender: i got the TE412P card. does the idea sound crazy you think? |
| 12:42.24 | hank | benjk: and even bristuff-stable wont work ootb if there are no kernel modules or they arent working. |
| 12:42.33 | hank | and _that_ is what i want to make sure. |
| 12:43.15 | hank | benjk: are you using misdn modules? |
| 12:43.18 | [TK]D-Fender | swytch: No its not crazy. Do you have an actual problem you are trying to diagnose? |
| 12:43.23 | benjk | the kernel modules are being built by the scripts that come with BRIstuff |
| 12:43.45 | benjk | no, BRIstuff comes with its own drivers |
| 12:43.49 | benjk | from Junghanns |
| 12:44.11 | benjk | and it dowloads and patches libpri for use with BRI |
| 12:44.59 | hank | i c, but i am trying to do this with debian and there asterisk-bristuff doesnt seem to contain modules. |
| 12:45.37 | hank | but ill remember that when i cant get it to work. thx for the hint :) |
| 12:46.08 | benjk | I use Ubuntu Server 6.06 |
| 12:46.18 | benjk | pretty much like Debian |
| 12:47.48 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
| 12:49.10 | hank | its based on debian... and if debian ships asterisk packages id really like to use them. |
| 12:51.25 | jamincollins | hank: Debian does ship asterisk packages. |
| 12:51.46 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 12:52.10 | hank | jamincollins: i know :) |
| 12:54.09 | hank | how do i test functionality of my isdn card and connection? |
| 12:54.11 | benjk | but without ISDN support that works out of the box or so it would seem |
| 12:54.26 | benjk | hook it up to an ISDN circuit |
| 12:54.30 | hank | it is |
| 12:54.41 | benjk | dial out |
| 12:54.50 | benjk | and another test would be to receive an incoming call |
| 12:55.43 | hank | uhm simplest way to dial out? |
| 12:57.07 | [TK]D-Fender | yay, new Polycom IP6xx serie w/ backlight and more expandability! |
| 13:00.24 | stoffell | [TK]D-Fender: awesome.. |
| 13:01.20 | [TK]D-Fender | stoffell: Indeed. Will badger my reseller for pricing and availability :) Its USB expandable in addition to other stuff... |
| 13:01.49 | stoffell | [TK]D-Fender: lol. and i only just start playing with the 430.. :) |
| 13:02.16 | [TK]D-Fender | stoffell: I had a busy weekend and didn't end up finishing the SIP 2.0.1 upgrade of my setup. Perhaps tonight. |
| 13:04.07 | stoffell | [TK]D-Fender: lol. same issue here.. downloaded but not installed yet, hehe |
| 13:04.39 | kmilitzer | Anyone else having tried to make a vanilla build for the latest zaptel 1.2.9? |
| 13:04.49 | stoffell | you know if it's normal that the polycom shows lost packets from time to time (menu->ethernet diagnostics) |
| 13:05.39 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 13:06.36 | pablus | morning |
| 13:07.34 | *** part/#asterisk k31th (n=keith@87.117.194.66) |
| 13:09.12 | hank | any ideas where to go for general support for isdn under linux? |
| 13:10.01 | [TK]D-Fender | stoffell: I never looked till just now and the one at my desk shows no lost packets |
| 13:10.42 | [TK]D-Fender | kmilitzer: There is a bug in the FTP tarball release. Download from SVN trunk. |
| 13:11.41 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 13:11.59 | kink0 | hello, I still having problems with deadlock :( anybody can help me ? |
| 13:12.21 | stoffell | hank: using bri cards? |
| 13:13.10 | hank | stoffell: i suspect its bri. its a billion usb adapter using a hfc chipset and the hfcsusb module from misdn. |
| 13:13.16 | hank | HFC-S USB: detected "Billion tiny USB ISDN TA 128" |
| 13:13.24 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.100) |
| 13:13.41 | RoyK | hank: i'd beleive the pci version might be better |
| 13:14.08 | *** join/#asterisk trelane` (n=trelane@unaffiliated/trelane) |
| 13:14.11 | stoffell | hank: brrrr usb |
| 13:14.17 | hank | RoyK: so do i |
| 13:14.30 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.42.141) |
| 13:14.34 | RoyK | and one third the price iirc |
| 13:14.35 | *** join/#asterisk prog (n=vdsoft@vdsoft.kh-net.cz) |
| 13:14.44 | prog | hello to * |
| 13:14.50 | kmilitzer | [TK]D-Fender: Is there already a bug filed for this? |
| 13:15.07 | kmilitzer | [TK]D-Fender: ... because I couldn't find one ... |
| 13:15.19 | symlink | yes there is |
| 13:15.21 | [TK]D-Fender | kmilitzer: I believe so. I ran into it last night and was warned by those working for Digium that its fixed in SVN |
| 13:15.26 | hank | stoffell: its an improvement nevertheless. i was supposed to work with an avm usb adapter before... |
| 13:15.38 | RoyK | kmilitzer: perhaps 1.2.12 will be a better choice |
| 13:15.44 | [TK]D-Fender | symlink: Will the tarballs be updated? |
| 13:15.46 | symlink | the release script couldn't deal with the way the Octasic firmware stuff |
| 13:16.02 | kmilitzer | [TK]D-Fender: OK ... as I don't need the latest zaptel, I will wait till a fixed tarball is released ... |
| 13:16.07 | symlink | yes |
| 13:16.14 | symlink | but it takes two people to make a new tarball |
| 13:16.17 | hank | stoffell, RoyK: can you tell me how to test functionality? |
| 13:16.44 | prog | is here anyone who is able helping me with call forwardings on asterisk ? I have some basics knowledge about it. Im able to store information about redirections in internal ASTDB, but my asterisk doesn`t use it when i make call. |
| 13:17.33 | prog | i dial *22*302 on 301 phone, but when I dial 301 on 302, its not redirected to 302 |
| 13:17.43 | RoyK | hank: it's simple dialplan logic |
| 13:17.54 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 13:18.03 | kmilitzer | RoyK: I don't understand why should asterisk 1.2.12 be a better choice than zaptel 1.2.9, these are two different things ;) |
| 13:18.16 | RoyK | kmilitzer: it is! lots! |
| 13:18.17 | RoyK | :) |
| 13:18.28 | RoyK | sorry. i misread |
| 13:18.47 | hank | RoyK: ah i see... no just kidding. i have no idea what youre talking about and no idea where i have to start here... |
| 13:18.49 | *** join/#asterisk Qwell (i=north@unaffiliated/qwell) |
| 13:18.49 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 13:19.29 | hank | all this linux and isdn stuff is a big question mark to me and all the reading i did couldnt noteworthy clear things up |
| 13:19.38 | kink0 | nobody has problems with deadlock in Asterisk ? |
| 13:19.47 | coppice | RoyK: you need a P4 machine if you want to make toast |
| 13:19.54 | RoyK | prog: exten => 293847238,1,Set(CALLERID(RDNIS)=${CALLERID(number)), exten => 293847238,n,Set(CALLERID(number)=${EXTEN}), exten => 293847238,n,Dial(Zap/g1/somenewnumber) |
| 13:20.02 | RoyK | hank: that one was for you |
| 13:20.10 | RoyK | coppice: true... |
| 13:20.29 | kmilitzer | Differnet question altogether: Any good ideas how to monitor SIP-Signaling of/with asterisk by using nagios? Right now I can check if asterisk is up and if I have enough free channels, but not if a call can be delivered ... |
| 13:20.31 | hank | RoyK: i see you have fun in building an even bigger question mark, right? ;) |
| 13:20.59 | RoyK | hank: that is three lines forwarding a call |
| 13:21.00 | RoyK | ~docs |
| 13:21.08 | jbot | rumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 13:21.09 | RoyK | ~book |
| 13:21.12 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:21.12 | RoyK | ~rtfm |
| 13:21.13 | jbot | rtfm is probably Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 13:22.28 | hank | wow thats some collection of documentation 'g' i hope it will clear things up and not push me deeper into confusion ;) ill be reading then... |
| 13:22.38 | hank | RoyK: thanks a lot :) |
| 13:23.17 | prog | RoyK: thank you, i will deeply check what you post me and try to exctract most relevant information from it ... thank you again |
| 13:24.56 | RoyK | prog: that was for hank. sorry. |
| 13:25.22 | prog | RoyK: yes, i see :-) |
| 13:25.49 | [TK]D-Fender | prog: * does not have any forwarding that you do put in yourself into extensions.conf. The is nothing stock/default/automatic/etc about it. |
| 13:27.09 | prog | [TK]D-Fender, i probably misundersood ... do you think that following line is bad ? : exten => _*22*XXX,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4 |
| 13:28.01 | RoyK | prog: pdepends what you want it to do :P |
| 13:28.34 | [TK]D-Fender | prog: Its missing "}) at the end but is otherwise sensible. Now you have to think "where the hell do I *USE* this value" |
| 13:29.33 | [TK]D-Fender | prog: exten => _*22*XXX,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4}) |
| 13:29.43 | prog | i need to make possible to my users to make redirection when they are on holidays. for example: user A with phone number 301 need to go away and redirect all incoming calls to his 301 phone to user B on phone 303 for example |
| 13:30.01 | prog | [TK]D-Fender, yes yes, a badly pasted , thank you for notice |
| 13:30.23 | [TK]D-Fender | prog: Well this is only the first part of that solution. Your extens that dial your phones need to sue this value to some practical end. By itself it means nothing. |
| 13:31.01 | *** join/#asterisk mercestes (n=merceste@216.54.143.242) |
| 13:31.38 | prog | [TK]D-Fender, you are right, EXTEN:4 number is just added into the database but never used ... and i do not know, how to use it- in other words - how to invoke the redirection |
| 13:31.44 | kmilitzer | prog: Why don't you use a mysql-DB or something like that and write a small AGI to enable/disable forwards .. |
| 13:32.13 | hank | hmpf i have seen most of that documentation already... almost useless atm. |
| 13:32.14 | kmilitzer | prog: And another agi to get the data back from the DB into asterisk? |
| 13:32.16 | [TK]D-Fender | prog: Before you dial your entensions device you should look it up and see if its redirected. Coding this is up to you. |
| 13:32.29 | prog | kmilitzer, because i had thought that ASTDB (internal berkley database ) is possible to use |
| 13:32.38 | hank | is there no _simple_ command line way to dial a number using an isdn device? or test whether it works? |
| 13:33.08 | kmilitzer | prog: Hmm, I don't know, ASTDB was always suspect to me, mysql is usable by multiple asterisk over IP ... |
| 13:33.22 | kmilitzer | hank: dial <number@context> |
| 13:33.36 | prog | kmilitzer, im not use what advantages are with mysql usage ... i have only 30-40 voip stations |
| 13:33.48 | hank | kmilitzer: what software package is dial in? |
| 13:34.02 | prog | <PROTECTED> |
| 13:34.12 | kmilitzer | hank: With command line you mean CLI, right? Or do you mean bash? |
| 13:34.29 | *** join/#asterisk littleball (n=littleba@cm201.omega152.maxonline.com.sg) |
| 13:34.44 | [TK]D-Fender | prog: Your idea is perfectly valid and the way I'd do it myself. Don't think about another DB method for this yet. |
| 13:35.00 | prog | [TK]D-Fender, thanks! |
| 13:35.12 | kmilitzer | As for bash do a # asterisk -rx "dial <number@context>" |
| 13:35.12 | hank | kmilitzer: a program, a script... id even use cat > myisdndevice <<EOF or whatever. |
| 13:35.14 | [TK]D-Fender | prog: You just need to make the extens that dial your phones check for forwarding before actually choosing WHO to call. |
| 13:35.29 | mercestes | kmilitzer: I've always had issues trying to run multiple *'s off a single database. |
| 13:35.32 | hank | kmilitzer: asterisk will have to know about my isdn device this way, right? |
| 13:35.33 | prog | [TK]D-Fender, could you advice me what is the syntax of these checks ? |
| 13:36.09 | kmilitzer | hank: Yes. And you need a working dialplan ... |
| 13:36.15 | kink0 | I get deadlock many frequenly :( pls help !!! |
| 13:36.26 | DarKnesS_WolF | there is no other way to create auto redial on busy ? just the .call file way ? |
| 13:36.58 | hank | kmilitzer: i _dont_ want to use asterisk yet. i simply want to test wether my system knows about my isdn device and if its working. |
| 13:37.01 | [TK]D-Fender | prog: the key things you'll need are "GotoIf" and "DB". |
| 13:37.49 | prog | [TK]D-Fender, thank you!!! i will check it ... please, is there some link where can i look for an example ? |
| 13:38.27 | prog | [TK]D-Fender, it seems im too lazy :-) ... but im working on it for several days but unsuccesfully |
| 13:39.01 | [TK]D-Fender | prog: Seeing as * was build with a collection of apps & functions in mind there is no "magic guide" for you. This is something you have to work out for yourself. Go to the WIKI for info on those 2 apps and see how they are used and try to imagine the conditions you are testing for before choosing how to dial. |
| 13:39.12 | kmilitzer | hank: Well, I am sorry, than I have no idea. I guess you can see, if you have a physical connection and/or your line is syncing (what kind of ISDN are you talking about anyway), but to make a call will be pretty complicated ... |
| 13:39.24 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
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| 13:40.14 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 13:40.31 | hank | kmilitzer: i dont want to make a call but someone suggested doing that to test the device. im probably talking about euroisdn. |
| 13:41.35 | *** part/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 13:41.39 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 13:41.41 | hank | i have some programs here like capiinit and capiinfo. should e.g. capiinit status tell me when the device is working? |
| 13:43.03 | stoffell | hank: what are you planning to use, capi, misdn or bristuff? (hfc) |
| 13:43.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:43.28 | stoffell | hank: capi only works for fritz cards (not for hfc for example) |
| 13:44.13 | hank | stoffell: 'grml.org' how do you know that? i didnt know i could use only one of those. |
| 13:45.25 | hank | stoffell: misdn is supposed to be the most up to date afaict so i think i want to use those. do you know any good isdn documentation? none of what ive read told me capi, misdn, and bristuff could not be used at the same time. |
| 13:45.50 | stoffell | hank: if it's a HFC chipser (your card), use misdn or bristuff. |
| 13:45.59 | *** join/#asterisk HaMYaI (i=CKGLOB@221.128.98.85) |
| 13:46.20 | stoffell | hank: it's not common practice to use more then 1 of those in 1 system, that's true. |
| 13:46.31 | hank | stoffell: why not? |
| 13:47.03 | HaMYaI | is it possible to see the established udp connections |
| 13:47.12 | stoffell | hank: because using capi and misdn is not something you normally do (you use OR capi-enabled cards OR hfc cards) |
| 13:47.17 | HaMYaI | netstat -an doesn't show them |
| 13:47.26 | boch | im looking for a SIP provider with an ASR > 70 to Bolivia/Paraguay/Peru, no matter rates, do know about one? |
| 13:47.40 | hank | HaMYaI: -anu? |
| 13:48.08 | hank | stoffell: hm ok ill probably use misdn since ive already compiled a kernel with those modules. |
| 13:48.31 | hank | stoffell: bristuff contains kernel modules as well? |
| 13:50.24 | HaMYaI | hank: that only shows the listening ones |
| 13:50.51 | HaMYaI | hank: or there's no established udp connection, anyone? |
| 13:52.35 | boch | there are no udp connections |
| 13:54.00 | HaMYaI | boc: ok, that's the answer |
| 13:55.18 | kink0 | well, I must go to lunch now, I hope I can get some help about deadlock later, thanks |
| 13:55.52 | stoffell | hank: bristuff has modules also, yes.. |
| 13:56.34 | hank | ok cool |
| 13:57.00 | *** join/#asterisk intralanman (n=lanman@pool-71-253-255-34.nrflva.east.verizon.net) |
| 13:57.46 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 13:57.48 | Bert- | hello there |
| 13:58.07 | Bert- | someone can tell me where can I get G729 codecs (want to test) |
| 13:58.18 | Bert- | I heard that I can use it to test (no for comercial use) |
| 13:58.20 | Qwell | Bert-: You buy them from Digium |
| 13:58.26 | Bert- | Qwell : I got one |
| 13:58.36 | Bert- | I have to call them because I already installed it |
| 13:58.39 | hank | im probably still pretty clueless 'g' |
| 13:58.40 | Bert- | (and my srv crashed) |
| 13:58.43 | hank | thats so frustrating |
| 13:58.52 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 13:58.56 | Bert- | but for now I don't want to register again on this host (it is not mine) |
| 13:59.09 | hank | linux and isdn is probably one of the most complicated topics... and i thought printing in linux was complicated. |
| 13:59.10 | Bert- | I just want to prove that it works well |
| 13:59.21 | Bert- | and I don't remember where to get that |
| 13:59.30 | Bert- | I know I event can test g723 but no need |
| 13:59.37 | Bert- | s/event/even |
| 14:00.41 | *** part/#asterisk HaMYaI (i=CKGLOB@221.128.98.85) |
| 14:01.43 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:03.45 | *** join/#asterisk Katty (n=Administ@64.82.232.54) |
| 14:03.54 | Katty | allo. |
| 14:04.14 | [TK]D-Fender | Katty: Salut ma chere ;) |
| 14:04.16 | *** join/#asterisk dr_chaos (n=dr_chaos@83.240.187.13) |
| 14:04.24 | dr_chaos | hi |
| 14:04.36 | dr_chaos | anyone familiar with sip trunking ? |
| 14:05.04 | swytch | [TK]D-Fender: well, we have problems of packet loss that we want to live with, and will see if the octassic chip can hide the problem for us. |
| 14:05.21 | Katty | [TK]D-Fender: morning, sunshine! |
| 14:05.27 | [TK]D-Fender | swytch: Packet-loss != EC..... |
| 14:05.36 | [TK]D-Fender | Katty: Mew. |
| 14:05.41 | Katty | [TK]D-Fender: think you can give me a paw? |
| 14:05.43 | [TK]D-Fender | Katty: Doin' OK here I guess.... |
| 14:05.54 | Katty | [TK]D-Fender:just okay? |
| 14:05.54 | [TK]D-Fender | Katty: Certainly |
| 14:05.55 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 14:05.59 | Katty | [TK]D-Fender: not peachy keen? |
| 14:06.02 | Katty | [TK]D-Fender: or just ducky? |
| 14:06.03 | swytch | [TK]D-Fender: yea, but octassic cointains "voice quality enhancement" in addition to ec. |
| 14:06.12 | [TK]D-Fender | Katty: Still a lot of life stuff to sort out. |
| 14:06.26 | Katty | [TK]D-Fender: aww, i'm sure you'll get through it all. |
| 14:06.31 | Katty | [TK]D-Fender: warning, not asterisk relate. |
| 14:06.35 | [TK]D-Fender | swytch: To think it reflects to SIP packet loss is more than a stretch, but ok. |
| 14:06.39 | Katty | [TK]D-Fender: but still not hard...i did it once, but didn't blog it. |
| 14:07.02 | Katty | [TK]D-Fender: my xfree86 config thinks my mouse is at /dev/input/mice. but it's not. i need to make a symlink to /proc/dev/whateverthemouse is. |
| 14:07.17 | Katty | [TK]D-Fender: cept, i don't know which one to cat to see if my mouse is on it. |
| 14:07.35 | Katty | [TK]D-Fender: it's a ps2 mousey, and i forgot how to make a symlink :< |
| 14:07.38 | swytch | [TK]D-Fender: because our network problems give us bad sound. also, some of the outgoing pstn operators (especially mobile) have bad sound to start with. |
| 14:07.40 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 14:07.54 | [TK]D-Fender | Katty: well you could try hard-coding the replacement as /dev/mouse I think.... give it a shot (commenting out the original. |
| 14:08.12 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 14:08.29 | swytch | [TK]D-Fender: we just thought: bad voice quality => should try "voice quality enhancement" +> lets try to force our traffic thru octassic. |
| 14:08.52 | Katty | [TK]D-Fender: but i still have to symlink it. |
| 14:08.55 | [TK]D-Fender | swytch: Like I said... a stretch. |
| 14:09.02 | Katty | [TK]D-Fender: which tty are ps2 mouseys generally on? |
| 14:09.16 | Katty | nevermind |
| 14:09.18 | Katty | i found it on psaux |
| 14:09.20 | swytch | [TK]D-Fender: do you think this "stretch" might work? |
| 14:09.25 | [TK]D-Fender | Katty: Don't really know this tech part so well :/ I merely "get by" with Linux actually |
| 14:09.40 | [TK]D-Fender | swytch: Doesn't make sense so the answer is "probably" ;) |
| 14:10.14 | swytch | [TK]D-Fender: i dont know exactly what is meant by "stretch" .. /-8 |
| 14:10.31 | hank | stoffell: why is there a mISDN_capi modules? |
| 14:10.57 | Katty | [TK]D-Fender: okie dokie. |
| 14:11.07 | Katty | anyone remember how to create a symlink? |
| 14:11.19 | [TK]D-Fender | swytch: These 2 items shouldn't have any impact on each other that I can think of, but who knows. just do what you want with it and see. |
| 14:11.26 | [TK]D-Fender | Katty: "man ln" |
| 14:11.30 | swytch | Katty: ln -s /full/path/source target |
| 14:11.33 | Katty | ln -s! |
| 14:11.37 | Katty | i remember now :P |
| 14:11.45 | [TK]D-Fender | Katty: I do know THAT much, just not the name of the actual device you'll want to target. |
| 14:11.46 | Katty | gotta stop drinking so much diet soda >.< |
| 14:11.51 | swytch | Katty: relative paths are dangerous |
| 14:12.01 | Katty | swytch: s'ok, i've done this before. |
| 14:12.09 | Katty | swytch: just gotta tell xserver where the mousey is |
| 14:12.50 | symlink | Kitty Katty! |
| 14:13.11 | swytch | [TK]D-Fender: what do you mean by "stretch"? you think its crazy? |
| 14:13.48 | Katty | symlink: allo. |
| 14:13.52 | [TK]D-Fender | swytch: What does SIP packet loss have to do with Echo Cancellation? You think EC can help cover that up? |
| 14:14.05 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:14.06 | symlink | Katty: I... tackle you and run away! |
| 14:14.59 | swytch | [TK]D-Fender: no i dont. |
| 14:15.46 | swytch | [TK]D-Fender: the wildcard we have does not only contain EC. it contains VQE as well. the latter may help us. thats what i will see. |
| 14:15.49 | [TK]D-Fender | swytch: Have you already bought the card? Do you actaully HAVE EC issue to be corrected sperate from your hopes of it impacting this other problem? |
| 14:17.01 | [TK]D-Fender | swytch: I'd find out what VQE actually means if I were you... |
| 14:17.03 | swytch | [TK]D-Fender: i dont know. if i have echo problems then i dont hear it at least. is it possible that i have it, but that it comes out as shit instead of echo because of the endpoint rtp trying to cancel it ??? |
| 14:17.49 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:18.46 | swytch | [TK]D-Fender: the delay in one of the two legs in a call is important, so we might have echo problems, but as i said. we dont hear echo, we hear shit. |
| 14:19.06 | puzzled | hi |
| 14:19.07 | [TK]D-Fender | swytch: SIP phones aren't supposed to EC anything. Thats your card/servers job |
| 14:19.36 | swytch | [TK]D-Fender: i have gateways as endpoints. not sip phones. |
| 14:20.01 | [TK]D-Fender | swytch: What kind? |
| 14:20.20 | *** join/#asterisk pacmanfan (n=a@dpc6935211186.direcpc.com) |
| 14:20.50 | swytch | [TK]D-Fender: audiocodes at callers end. an echo would have been heard as the delay is important. |
| 14:22.29 | [TK]D-Fender | swytch: Well if its an FXS gateway it shouldn't be doing any EC. |
| 14:22.59 | *** join/#asterisk WaysFuder (i=WaysFude@Conexao.Segura.Nikity.133.PredialNET.com.br) |
| 14:23.04 | swytch | [TK]D-Fender: anal_phone ---> AUDIOCODES-118fxs_sip ---> sip_ASTERISK_sip ---> sip_MERA_sip ---> remote_pstn_gateway |
| 14:24.12 | swytch | [TK]D-Fender: on the branch audiocodes to asterisk the ping sais 50ms max |
| 14:24.16 | WaysFuder | what's event to get if SIP linked to ZAP? |
| 14:24.45 | WaysFuder | I'm developing integrated solution with Delphi App. |
| 14:25.11 | swytch | [TK]D-Fender: audiocodes fxs _has_ EC. |
| 14:26.15 | dr_chaos | [TK]D-Fender: icmp ping can't measure the real latency on the lines, expecially if you use internet tunnels |
| 14:26.58 | pacmanfan | i hate asking here an a * channel, but i can't find any PBX users elsewhere... anyone around that's familiar with Partner systems? |
| 14:27.08 | swytch | [TK]D-Fender: ok, its adsl pluss 17 inet hops. how can i mesure without icmp? |
| 14:27.54 | kink0 | I need help with deadlock, my asterisk uptime is not longer than about 30 minutes :( |
| 14:31.16 | *** join/#asterisk Bambr (n=Bambr@213-35-232-7-dsl.end.estpak.ee) |
| 14:33.33 | dr_chaos | swytch: are u using unix or windows ? |
| 14:33.43 | swytch | dr_chaos: *nix |
| 14:34.08 | swytch | dr_chaos: windoze adds more delay? |
| 14:34.13 | dr_chaos | what dist ? |
| 14:34.28 | dr_chaos | nao windows ping uses udp frames not icmp |
| 14:34.35 | swytch | dr_chaos: debians gnu/linix |
| 14:35.26 | dr_chaos | i checking for package that have a udp or tcp ping |
| 14:37.16 | swytch | dr_chaos: ah ok. i have one "client" site with a debian too, so i would be able to effectuate a test over the inet with two debians yes. |
| 14:37.52 | swytch | dr_chaos: but what do you think about my "hidden echo problem" ? |
| 14:38.09 | dr_chaos | there is a bandwith tool called iperf |
| 14:39.35 | swytch | the "client" site machine may be heavily loaded. is that a problem for the tests outcome usability? |
| 14:40.13 | *** join/#asterisk Qwell (i=north@unaffiliated/qwell) |
| 14:40.13 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:40.55 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:40.56 | dr_chaos | swytch: test echoping |
| 14:41.10 | kink0 | I feel like alone in the world with deadlock problems , appears nobody more suffers of this crashes. |
| 14:41.12 | swytch | [TK]D-Fender: what do you think about my "hidden echo problem" ?? |
| 14:41.26 | swytch | dr_chaos: iperf and echoping, ok. thank you. |
| 14:41.47 | dr_chaos | a mrtg mon. you lines would be perf |
| 14:41.48 | [TK]D-Fender | swytch: Well you described it as "noise" and the fact its PL related I think your networking scenario may simply not be terribly viable... |
| 14:42.00 | swytch | dr_chaos: they both mesure delay too. without being fooled by a too hevily loaded host? |
| 14:42.29 | kink0 | I have try to upgrade my zaptel and asterisk, but while compiling zaptel I got "oct6100api/oct6100_api.h', needed by `/usr/src/zaptel-1.2.9/wct4xxp/vpm450m.o" and compilation Error, any help about what is this ? |
| 14:42.30 | tzanger | [TK]D-Fender: do you know if Bell has "universal" milliwatt numbers? |
| 14:42.37 | tzanger | 958-2522 seems to be the universal ANI |
| 14:42.43 | swytch | [TK]D-Fender: whats "PL related" ?? |
| 14:43.22 | tzafrir | [TK]D-Fender, the EC should be done as close as possible to the analog phone |
| 14:43.22 | tzanger | er 2622/2422 |
| 14:43.35 | dr_chaos | swytch: a heavy loaded host generates delay in codecs that you may think is delay |
| 14:43.41 | dr_chaos | echo |
| 14:43.49 | WaysFuder | what's event to get if SIP linked to ZAP? |
| 14:43.51 | dr_chaos | swytch: a heavy loaded host generates delay in codecs that you may think is echo |
| 14:44.01 | [TK]D-Fender | swytch: Packet Loss |
| 14:46.26 | swytch | dr_chaos: the machines involved in the rtp are not at all heavy loaded, it was talking about the debian machine at one of the end that i would use to do an echoping test. |
| 14:47.09 | swytch | /it/i/ , yes i have the habit of talking about myself in third person. |
| 14:47.36 | *** join/#asterisk ukris (n=ukris@p6215-ipad96marunouchi.tokyo.ocn.ne.jp) |
| 14:47.43 | *** join/#asterisk Sloboda (n=slob@local.proggart.com) |
| 14:48.14 | swytch | s/third person/third computer/ |
| 14:48.33 | Sloboda | Hi! Is the way for the member of the queue to choose what caller reply to? |
| 14:49.02 | pacmanfan | no Partner users around? |
| 14:49.04 | Sloboda | Using Manager API or something else? |
| 14:49.19 | WaysFuder | nick WaysFuder[Brazil] |
| 14:49.21 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:50.14 | swytch | [TK]D-Fender: but if we are going to live with a little bit of PL, then maybe the VQE can diminish the noice .. |
| 14:53.13 | [TK]D-Fender | swytch: Give ti a try I guess.... |
| 14:53.19 | *** join/#asterisk kph100 (n=kph100@206-248-156-72.dsl.teksavvy.com) |
| 14:54.22 | swytch | [TK]D-Fender: yes. just bad that the TE412P was in wrong PCI format for our computer. |
| 14:56.08 | symlink | [TK]D-Fender: moo |
| 14:56.09 | *** join/#asterisk evosys (n=root@194-105-181-29.ifb.co.uk) |
| 14:56.35 | *** join/#asterisk volp (n=volp@201.210.82.39) |
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| 14:58.57 | MrChimpy | AMI can't cope with my call load. When I'm doing about a few hundred calls/min with event monitoring on it just drops the AMI connection |
| 14:59.00 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:01.10 | *** part/#asterisk volp (n=volp@201.210.82.39) |
| 15:02.13 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.166) |
| 15:05.52 | [TK]D-Fender | swytch: Sangoma for the Win! |
| 15:05.55 | [TK]D-Fender | :O |
| 15:06.00 | [TK]D-Fender | symlink: *poke* |
| 15:06.06 | symlink | yes'm |
| 15:06.31 | MrChimpy | frikkin dumb digium cards. yay sangoma. |
| 15:06.51 | symlink | lol |
| 15:10.13 | *** join/#asterisk ToyMan (n=stuq@74-32-59-182.dsl1.mdl.ny.frontiernet.net) |
| 15:10.27 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:12.16 | *** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net) |
| 15:12.20 | Bert- | hello again |
| 15:12.35 | Bert- | about that : Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 15:12.42 | *** join/#asterisk ToxaP (n=ToxaP@213.227.193.75) |
| 15:12.59 | Bert- | how to know on which side this extra frame comes from please ?? |
| 15:13.24 | Bert- | hmm ym english is pretty strange :) |
| 15:14.17 | eKo1 | How many G.729 calls are you handling? |
| 15:14.45 | *** join/#asterisk ToxaP (n=ToxaP@213.227.193.75) |
| 15:15.46 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
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| 15:16.33 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 15:17.06 | *** join/#asterisk jbroome (n=jbroome@unaffiliated/jbroome) |
| 15:17.25 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr) |
| 15:17.31 | *** join/#asterisk Vec (n=Vector@dsl-146-119-138.telkomadsl.co.za) |
| 15:18.12 | Bert- | only one |
| 15:18.14 | jhiver | Has ANYBODY managed to have reliable and scalable H323 <-> SIP translation with open source software? I've been working on this for weeks with no results |
| 15:18.26 | Bert- | With a thomson 2030 |
| 15:19.04 | *** join/#asterisk anthonyl (n=anthonyl@c-67-167-214-149.hsd1.il.comcast.net) |
| 15:19.13 | jbroome | Is there a way to get polycoms to dump their current sip.cfg file to the ftp/tftp server? I need to disable the MWI beep, and don't want to redo my configs on all my phones |
| 15:19.52 | eKo1 | Bert-: Great. Now you know where that message is comming from. |
| 15:20.18 | Bert- | eKo1! |
| 15:20.21 | eKo1 | jhiver: What do you mean by translation? |
| 15:20.21 | Bert- | there is two channels |
| 15:20.28 | Bert- | from my sip phone |
| 15:20.37 | Bert- | and the remote part (sip proxy) |
| 15:20.40 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:20.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:20.43 | Bert- | s/is/are |
| 15:20.44 | jhiver | means being able to receive H323 calls and foward them as SIP, and vice versa |
| 15:20.46 | *** join/#asterisk j4m3s (i=j4m3s@nat/digium/x-3bf7c40d2188de92) |
| 15:21.10 | Bert- | I don't know if it mistake is from my phone or from remote side |
| 15:21.15 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:21.19 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:21.37 | jhiver | chan_h323 has got "lock" issues, plus it doesn't send early RTP when receiving calls |
| 15:21.47 | eKo1 | Bert-: Turn off VAD on the phone. If that doesn't do it, then you know the problem is on the other side. |
| 15:21.56 | jhiver | Yate i've spent the afternoon trying to do an interconnect with, with a paid consultant, no problem |
| 15:22.04 | eKo1 | jhiver: I've never had lock issues with chan_h323. |
| 15:22.42 | jhiver | oh323 does'nt handle more than like 5 channels |
| 15:22.42 | jhiver | and ooh323 i'm trying to compile it now |
| 15:22.42 | kph100 | i need to create contexts dynamically using realtime extensions. How can I do this?? |
| 15:22.43 | eKo1 | I've handled way over 5 channels. |
| 15:22.43 | jhiver | I did, above around 20 channels simultaneous |
| 15:22.49 | jhiver | the ASR on the gateways was around 40%, with my clients softswitch it was around 70% |
| 15:23.04 | *** part/#asterisk j4m3s (i=j4m3s@nat/digium/x-3bf7c40d2188de92) |
| 15:25.16 | jhiver | eKo1, you're obviously in better luck than i am |
| 15:25.24 | jhiver | which versions are you using? |
| 15:25.47 | eKo1 | Of what? |
| 15:25.55 | phearless | how can I set the same Directory for all my SPA (Sipura) phones ? |
| 15:25.59 | jhiver | asterisk / chan_h323 |
| 15:26.18 | jhiver | also is actually ooh323 any good / usable? |
| 15:26.42 | eKo1 | Asterisk 1.2 with chan_h323 from channels/h323 |
| 15:26.44 | *** join/#asterisk nortex (n=nortex@ppp-70-252-53-238.dsl.amrltx.swbell.net) |
| 15:26.46 | sivana | can 'maxmsg' be set per mailbox in voicemail? |
| 15:26.46 | jhiver | latest CVS doesn't seem to compile, i'm trying to compile 1.2 now |
| 15:26.56 | eKo1 | I have never bothered with ooh323 |
| 15:27.00 | jhiver | eKo1, that's what i'm using too |
| 15:27.22 | *** join/#asterisk oej (n=oej@64.251.112.98) |
| 15:27.32 | jhiver | eKo1, are you building from scratch or using package? |
| 15:27.44 | jhiver | from voip-build? |
| 15:27.50 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 15:28.53 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:31.25 | *** join/#asterisk Remenic (n=remenic@konversation/user/Remenic) |
| 15:31.44 | Remenic | hi, is it possible to force a hangup from the asterisk console? |
| 15:32.18 | Remenic | oh, I think the answer is hangup... why didn't I try that.. |
| 15:32.44 | Remenic | oh that only hangs up calls placed from the console |
| 15:32.54 | Remenic | but I want to hangup ALL lines :) |
| 15:33.15 | phearless | how can I set the same Directory of name/exten for all my SPA (Sipura) phones ? |
| 15:33.25 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 15:33.28 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:33.28 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:33.57 | syzygyBSD | how can I limit the number of outgoing calls on a sip extension? |
| 15:34.30 | [TK]D-Fender | phearless: What directory are you talking about? |
| 15:34.35 | syzygyBSD | I updated it to outgoinglimit=10 but and reloaded but it still tries to dial all 20 out |
| 15:35.47 | phearless | [TK]D-Fender: the list of name of people and their phone number |
| 15:36.19 | syzygyBSD | does it matter that the remote side hasn't picked up yet? |
| 15:36.55 | [TK]D-Fender | phearless: :What/what is this magic list you are talking about? |
| 15:39.18 | phearless | a list to avoid to know by heart all the exten |
| 15:39.22 | *** join/#asterisk |Vulture| (n=Vulture@101.222.121.70.cfl.res.rr.com) |
| 15:39.39 | phearless | you push the button "dir" and you got a list of phone numbers |
| 15:39.59 | |Vulture| | Anyone here ever install a PRI from a Cisco VWIC-1MFT-T1 card? |
| 15:40.02 | phearless | for example the Cisco 7960 allow to use a remotely hosted directory |
| 15:40.06 | |Vulture| | All I am receiving is a red alarm |
| 15:40.18 | phearless | but for the SPA-942 I can't find how to do it |
| 15:41.39 | phearless | [TK]D-Fender: make sense ? |
| 15:41.51 | syzygyBSD | so in 'sip show peer NAME' it shows I havve a call limit of 10, why do 20 calls go out then? |
| 15:42.23 | [TK]D-Fender | phearless: Ah on a SIP hardphone. I take it you're talking about the SPA-841 since thats the only one made in their name? |
| 15:42.49 | [TK]D-Fender | phearless: So that'd be Linksys..... Have you read the admin guide? |
| 15:42.57 | phearless | http://voipspeak.net/index.php?option=com_content&task=view&id=71&Itemid=27 |
| 15:43.09 | phearless | Linksys SPA-942 SIP Phone |
| 15:43.33 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
| 15:43.46 | afrosheen | hooyah does anyone have experience fixing a broken FOP config? |
| 15:44.12 | phearless | I have not found the admin guide of the SPA-942, only the SPA841AdminGuide0105-DRAFT.pdf |
| 15:44.38 | *** join/#asterisk momelod (n=momelod@HSE-Montreal-ppp143067.sympatico.ca) |
| 15:45.23 | momelod | hey people |
| 15:46.06 | momelod | can anyone give my an idea why i experience echo only when i initiate the call from sip->zap->pstn but no echo when the call comes to me pstn->zap->sip |
| 15:46.08 | [TK]D-Fender | phearless: Look harder - http://www.voip-info.org/wiki/view/SPA-941 |
| 15:46.10 | afrosheen | bring some brains, we got plenty of zombies today |
| 15:46.16 | phearless | oh there is a SPA941AdminGuide.pdf |
| 15:46.37 | phearless | [TK]D-Fender: I will have a look at this admin guide :) |
| 15:47.20 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:50.09 | [TK]D-Fender | phearless: I don't see the common directory as an option in there. It may simply not be possible. |
| 15:50.33 | phearless | there is a "DirEntry", I will try this |
| 15:51.46 | *** join/#asterisk kratzers (n=kratzers@kratzers.static.pa.net) |
| 15:52.11 | [TK]D-Fender | phearless: That appears to be distinctive ringing by CID |
| 15:52.35 | [TK]D-Fender | phearless: Oh well. |
| 15:52.48 | phearless | I try, I will let you know if it worked |
| 15:53.30 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:54.16 | jhiver | is the latest ooh323 only supporting ulaw & gsm still? |
| 15:54.30 | jhiver | cause that would make it pretty useless (for me) |
| 15:59.29 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 16:00.54 | caio1982_ | does someone here knows a tool for encoding .wav files into g729 compatible audio? (for linux if possible, but it's ok if it's for windows) i'm looking for someone like that to avoid transcoding in some specific scenarios |
| 16:00.59 | *** join/#asterisk jaike (i=jaike@210.5.116.99) |
| 16:01.09 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 16:01.52 | [TK]D-Fender | caio1982_: That IS transcoding... the fact you record the otput for future direct streaming is besides the point. You'd need a license to convert it in the first place. |
| 16:02.33 | caio1982_ | [TK]D-Fender: i own a bunch of licences, that's not a big problem in my case. the missing tool actually is :/ |
| 16:02.39 | Jason99 | I am running two Asterisk servers and I am having problems with DTMF. When server A dials Server B, Server B prompts with dial tone using DISA. When I enter digits, the dial tone continues and it does not detect the DTMF. Server A/B are both configured as rfc2833. Can someone point me in the right direction on how I can troubleshoot this? |
| 16:02.52 | caio1982_ | [TK]D-Fender: the idea is to avoide transcoding at call time, but encoding it previously |
| 16:03.05 | afrosheen | caio1982_: I thought sox did that |
| 16:03.17 | caio1982_ | g729? didn't know that |
| 16:03.35 | caio1982_ | lemme check |
| 16:03.48 | momelod | so no one else has experienced one-way echo? |
| 16:04.07 | momelod | i only get echo when i dial out.. |
| 16:04.14 | momelod | incomming calls are perfect |
| 16:04.15 | jaike | 1.2.12 seems to be pretty stable with mixmonitor |
| 16:04.35 | file | jaike: no issues? |
| 16:04.39 | afrosheen | oh I doubt it'll transcode to g729, could be wrong |
| 16:04.41 | caio1982_ | afrosheen: "PCM, U-law, A-law, G7xx ADPCM files" hmmm G7xx is a little big generic for me... i'll take a look at how sox handle it |
| 16:04.46 | jaike | file: 8 hours, 50 simul calls, stil up |
| 16:04.57 | file | jaike: excellent. |
| 16:05.00 | afrosheen | if you have licenses, maybe |
| 16:05.00 | jaike | 1.2.11 crashes after an hour |
| 16:05.05 | jaike | hope this keeps up |
| 16:05.18 | caio1982_ | afrosheen: i do :) thanks for the remind |
| 16:05.18 | jaike | nice work file |
| 16:05.19 | caio1982_ | about sox |
| 16:05.28 | file | you're the first one who has said "yes it works great" |
| 16:05.31 | file | everyone else has been silent |
| 16:05.50 | jaike | file: hehe..still keeping my fingers crossed though |
| 16:05.57 | afrosheen | 1.2.11 works beautifully here |
| 16:06.05 | *** join/#asterisk RoyK (n=roy@ti211210a080-0574.bb.online.no) |
| 16:06.52 | jaike | afrosheen: the crashes usually happen during heavy loads, 30 simultaneous calls and up |
| 16:07.16 | afrosheen | jaike: we have alot of extensions but don't have the bandwidth for that kind of load..so I haven't seen that |
| 16:07.26 | jaike | i can run 1.2.11 with 10 calls and nothing happens |
| 16:07.37 | file | it requires certain timing to happen |
| 16:07.48 | afrosheen | is it zap related? |
| 16:07.51 | file | no |
| 16:07.55 | afrosheen | nasty |
| 16:07.58 | jaike | file: but i was able to crash it with 1 call. i couldnt replicate it again |
| 16:08.06 | afrosheen | hmm |
| 16:08.07 | Bert- | is background() able to deal with .gsm files ? |
| 16:08.07 | file | let me rephrase... it requires events to happen in the right order |
| 16:08.08 | jaike | afrosheen: were doing pure voip |
| 16:08.15 | file | if it's that order, it goes KABOOM |
| 16:08.23 | afrosheen | jaike: we are too but we have a zap card for conference timing |
| 16:08.28 | afrosheen | wondered if that had any bearing on it |
| 16:08.42 | jaike | i have ztdummy loaded |
| 16:08.47 | jaike | for meetme |
| 16:08.54 | jaike | dunno if that has any effect |
| 16:08.59 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 16:09.02 | afrosheen | it might be worth investigating |
| 16:12.00 | jaike | will report tomorrow if anything happens. cant go back to monitor anymore, utilization stays 100% when we use it. with mixmonitor, were toping 35% |
| 16:14.08 | file | jaike: I might be on/off tomorrow, but send me a private message and I really would like to know how it goes |
| 16:14.21 | jaike | file: ok, will do |
| 16:15.27 | vader-- | question: i have some older phones that are hooked into a digium tdm2400p card and they have message indicator lights on them |
| 16:15.27 | vader-- | is there anyway to get them to work? |
| 16:15.32 | *** part/#asterisk jaike (i=jaike@210.5.116.99) |
| 16:16.26 | afrosheen | vader--: I imagine so but I don't know the answer |
| 16:16.55 | vader-- | ya i would think you could get them to work |
| 16:16.57 | vader-- | but im unsure |
| 16:16.59 | vader-- | as well |
| 16:18.21 | phearless | DirENNtry does not seem to work , [TK]D-Fender :( |
| 16:18.42 | phearless | DirEntry* |
| 16:19.29 | *** join/#asterisk schotten83 (n=stefano@200.138.30.10) |
| 16:19.57 | schotten83 | hey m8, does anyone know that if Intel 537EP is working with latest zaptel drivers? |
| 16:20.13 | schotten83 | @ voip-info says that dont, but maybe that is outdated |
| 16:22.08 | afrosheen | schotten83: you can always try building the latest and loading the module right |
| 16:22.18 | Jason99 | Hey anyone familiar with DISA? two servers, when server A dials Server B, Server B prompts with dial tone using DISA. When I enter digits, the dial tone continues and DTMF not detected. Server A/B are both configured as rfc2833. Can someone point me in the right direction on how I can troubleshoot this? |
| 16:22.53 | [TK]D-Fender | phearless: Its purpose was pretty apparent from the guide. Guess you're SOL... |
| 16:23.11 | afrosheen | Jason99: you can enable sip debug or just debug and tail -f /var/log/asterisk/full to watch the call progress |
| 16:24.14 | phearless | [TK]D-Fender: SOL ? |
| 16:24.31 | afrosheen | shit outta luck |
| 16:24.31 | phearless | Sadly Outta Luck ? |
| 16:26.38 | [TK]D-Fender | ^^ |
| 16:29.50 | *** join/#asterisk hobbs (n=andrew@p3m/member/hobbs) |
| 16:33.04 | schotten83 | afrosheen ty, but Im going to buy the card, so I would like to know if thats valid before I buy it |
| 16:36.13 | *** join/#asterisk angom (n=angom@red-corp-200.79.129.196.telnor.net) |
| 16:36.17 | Jason99 | afrosheen: it was because the phones themselves were set for inband |
| 16:36.24 | Jason99 | thanks |
| 16:38.49 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 16:39.49 | *** join/#asterisk stubert (i=stu@techtools.actusa.net) |
| 16:44.06 | [TK]D-Fender | schotten83: Perhaps you should just buy a supported card.... |
| 16:44.44 | *** join/#asterisk mivck (i=1000@200.114.70.228) |
| 16:47.23 | caio1982_ | caio1982@caio:/tmp/sox-12.17.9$ grep -ir g729 * |
| 16:47.23 | caio1982_ | caio1982@caio:/tmp/sox-12.17.9$ ls |
| 16:47.26 | caio1982_ | afrosheen: ^ |
| 16:47.28 | caio1982_ | afrosheen: hehe |
| 16:48.06 | *** join/#asterisk Givemelove (n=foo@208.57.229.162) |
| 16:48.43 | afrosheen | Jason99: interesting |
| 16:48.52 | afrosheen | Jason99: what phones were they? |
| 16:48.55 | *** join/#asterisk ramtha (n=tk@p5088B67B.dip0.t-ipconnect.de) |
| 16:48.58 | ramtha | heyya |
| 16:49.51 | afrosheen | caio1982_: http://www.asteriskguru.com/audio_conversion.php |
| 16:50.04 | caio1982_ | afrosheen: thanks |
| 16:50.37 | ramtha | can i hand a Dial cmd rule from one xtension to an other. i wnat to GoTo jump from one context to an other. in the first context, i wnat to hand the |60|tT to the new extension. is this possible? |
| 16:51.39 | [TK]D-Fender | ramtha: Set a variable with it or something. |
| 16:52.42 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 16:52.56 | ramtha | Dial SIP/${EXTEN}|$Var |
| 16:53.04 | ramtha | does this work? |
| 16:53.43 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 16:57.06 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 16:57.31 | *** join/#asterisk oej (n=oej@64.251.112.98) |
| 16:58.12 | *** join/#asterisk Avalone (n=Avalone_@dial-344.vl-cen-as1.avtlg.ru) |
| 17:00.12 | [TK]D-Fender | ramtha: When you clean up your formatting, sure |
| 17:00.13 | *** join/#asterisk beu_ (i=beu@freenode/developer/gentoo.developer.beu) |
| 17:03.59 | Ryushin | Is it possible, through the polycom phones, to find out if someone is on the phones before transferring a call to them? |
| 17:04.42 | afrosheen | Ryushin: shared call appearance? FOP will show you..if you feel like hassling with getting it installed |
| 17:05.06 | kratzers | attended transfer |
| 17:05.21 | Ryushin | It's been requested. What is FOP? |
| 17:05.29 | afrosheen | flash operator panel |
| 17:05.41 | afrosheen | you can do transfers, see who has voicemail, etc. |
| 17:06.08 | Ryushin | Is it part of asterisk? |
| 17:06.14 | ramtha | can i use the pickup/callergroup with indication a call directly a phone (led or something) in the group? |
| 17:06.28 | afrosheen | nope, it snaps into asterisk, google it, it's cool if you can get it working (not that hard) |
| 17:06.29 | CunningPike | Ryushin: Why not use presence? |
| 17:06.43 | ramtha | if the incoming phone call isnt heart, i do not know that someone in my group is called |
| 17:06.53 | Ryushin | I haven't heard of either one. What is Presence? |
| 17:07.28 | RoyK | order |
| 17:07.46 | CunningPike | Ryushin: If you create hints for your SIP extensions, you can set up presence monitoring (called buddies in Polycom-land) to have them indicate when they're busy |
| 17:07.53 | afrosheen | Ryushin: http://www.asternic.org/ |
| 17:08.04 | mut | so how about this idea.. i buy a lucent excel switch, sick asterisk in there with 2 quad pri cards, 4 pri to take calls in so i can bill calls, and 4 pri out so i can do callerid name |
| 17:08.40 | Ryushin | Okay, I'll go dig into both. Thanks guys. |
| 17:08.51 | afrosheen | have fun with it |
| 17:10.55 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 17:11.28 | *** join/#asterisk dserban (n=dserban@caliban.lodgingcompany.com) |
| 17:11.31 | *** join/#asterisk niter3 (i=niter3@pr0tected.us) |
| 17:12.07 | Ryushin | afrosheen: FOP looks very nice. Kinda more what they are looking for. Too bad the demo is down. |
| 17:12.12 | flujan | hi guys... I am having problems with realtime queues... If i have a realtime queue with musiconhold... the musicon hold just not work. |
| 17:12.21 | afrosheen | Ryushin: trust me, it's good people, you will like it :) |
| 17:12.33 | afrosheen | Ryushin: how many extensions are you supporting? |
| 17:12.39 | flujan | if I have a static queue in queues.conf The musiconhold plays well. :( |
| 17:13.06 | dserban | Can someone please explain the wonderful intricacies of ael? I have an extensions.conf which defines the [general] context, then my extensions.ael handles the rest. pbx_ael.so is being loaded... and I just get a simple pbx.c:3761 ast_merge_contexts_and_delete: Requested contexts didn't get merged error. Which means crap to me. Can someone please give me a clue? |
| 17:13.30 | Ryushin | afrosheen: About 25 |
| 17:13.54 | flujan | Anyone working with realtime queues and postgresql? |
| 17:13.58 | Zodiacal | anyone use asterfax? is there a way to set the retry number and times? |
| 17:14.13 | dserban | No. I'm trying to actually get queues to work without chan_agent ... |
| 17:14.46 | [TK]D-Fender | Ryushin: Polycom IP 6XX series phones are great with the Attendant modules lit up like a Christmas tree :) |
| 17:15.09 | *** part/#asterisk hobbs (n=andrew@p3m/member/hobbs) |
| 17:15.20 | Zodiacal | nm |
| 17:16.55 | *** join/#asterisk HaMYaI (i=CKGLOB@221.128.98.85) |
| 17:17.10 | Ryushin | [TK]D-Fender: Yea, I got one 601 with two extra attendant modules. I haven't set those up yet. That is for the main secretary, but they have the main line ring three other phones in the office, and for when the secretary isn't available, they'd like to know who is on what. |
| 17:17.46 | Ryushin | I'll be setting up the attendant modules and probably FOP tomorrow. It will be fun. |
| 17:18.02 | flujan | holy smokes... I dunno why I hear no sound since i have this: http://pastebin.ca/167083 |
| 17:18.28 | flujan | The realtime queue start music on hold just like the static one... but I can hear nothing... |
| 17:18.40 | flujan | I am freeze... No idea why is it happening. |
| 17:19.37 | syzygyBSD | is 255 valid in ip addresses? not masks or broadcasts, but real ips |
| 17:19.45 | syzygyBSD | ie 72.255.2.3 |
| 17:20.27 | syzygyBSD | I guess if the subnet is outside that range it would work... |
| 17:21.16 | inv_arp[work] | syzygyBSD: correct |
| 17:21.20 | *** join/#asterisk fafnir (n=notfaf@unaffiliated/fafnir) |
| 17:22.05 | *** join/#asterisk oej (n=oej@64.251.112.98) |
| 17:22.47 | fafnir | anyone seen shido6 around? |
| 17:26.40 | [TK]D-Fender | Ryushin: Many ways to do that. I have a MicroBrowser scrip for all my 600 series as well |
| 17:29.22 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 17:31.37 | *** join/#asterisk pollohawk (n=pollohaw@mmail.picksend.com) |
| 17:33.09 | *** join/#asterisk JohnRobert (n=rg@i-195-137-116-41.freedom2surf.net) |
| 17:34.14 | *** join/#asterisk joako (n=kvirc@64.238.172.235) |
| 17:34.17 | pollohawk | Has anybody got pwlib_Mimas_patch2 to compile on a 2.6.15 amd64 kernel |
| 17:34.24 | syzygyBSD | does sip use all the bandwidth for ulaw if the call isn't answered yet? |
| 17:34.36 | JohnRobert | I've bought two atcom 168V ata's...they're from china, and I've tried hooking my phone up, but I get no sound or anything on the handset...I was wondering if I needed an adapter for them as I'm in the uk and use uk handsets. |
| 17:34.50 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 17:34.50 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 17:34.59 | FuriousGeorge | hry all |
| 17:35.02 | [TK]D-Fender | syzygyBSD: If the call hasn't been answered then there is no RTP. SIP itself takes up virtually no traffic |
| 17:35.03 | FuriousGeorge | *hey |
| 17:35.06 | JohnRobert | is it normal to need an inverter or something for uk handsets? |
| 17:35.23 | joako | John: A standard telephone should work.... make sure you are using the correct pin-out and that the box is configured and registered |
| 17:35.31 | SplasPood | Anyone here have a Polycom IP 4000 (conference room phone) ? |
| 17:35.40 | [TK]D-Fender | JohnRobert: Whatever you would require to use a UK phon on a North American jack. |
| 17:35.53 | [TK]D-Fender | SplasPood: One of my clients does, whats your question? |
| 17:36.01 | syzygyBSD | [TK]D-Fender: ok, thanks, what if I had a custom sound being streamed back to the calling phone? does it only take 1/2 the bandwidth? |
| 17:36.12 | syzygyBSD | or does that not work |
| 17:36.25 | FuriousGeorge | does anyne know why i can start asterisk with asterisk -c but safe asterisk immediately dies. It appears that when i use safe_asterisk it cant find the zap modules for my tdms |
| 17:36.31 | SplasPood | [TK]D-Fender: Just took one out of the box... Interestingly it seems that if I dial.... 1+AC+Number it actually ends up dialing 1+1+AC+Number... Even tho it's using the same configs as my 501s and 601s |
| 17:36.38 | SplasPood | [TK]D-Fender: Haven't investigated yet... |
| 17:36.39 | [TK]D-Fender | syzygyBSD: All or northing. thats it. |
| 17:37.01 | syzygyBSD | ok, thanks |
| 17:37.26 | JohnRobert | I'm using one of those laptop modem cables to connect the phone to the ata...could this be the problem? |
| 17:37.48 | JohnRobert | the ata is on and configured |
| 17:37.57 | [TK]D-Fender | JohnRobert: Is it appriate to use that cable with a UK phon directly into a N/A jack? |
| 17:38.08 | syzygyBSD | ok, so should a 3.2Ghz pentium be able to handle a 23 calls of Zap -> g729 Sip? |
| 17:38.37 | syzygyBSD | my testing suggests it is a bit lower, 15 calls |
| 17:38.46 | [TK]D-Fender | syzygyBSD: More than easily I would say |
| 17:38.48 | SplasPood | [TK]D-Fender: ever seen that? |
| 17:38.49 | JohnRobert | [TK]D-Fender, I'm not sure, that's why I'm asking :p |
| 17:39.15 | [TK]D-Fender | JohnRobert: Thats a general telco question. Ask your equivalent to Radio Shack. |
| 17:39.23 | JohnRobert | maplin :p |
| 17:39.38 | JohnRobert | they probably won't know :/ |
| 17:39.49 | [TK]D-Fender | JohnRobert: Thats sad... |
| 17:40.09 | JohnRobert | I feel this atcom ata was a poor purchase |
| 17:40.17 | JohnRobert | it's all in badly translated chineese |
| 17:40.21 | FuriousGeorge | i cant be entirely sure why its crashing either because nothing is being written to the logger |
| 17:40.21 | JohnRobert | chinese |
| 17:40.32 | [TK]D-Fender | JohnRobert: It is cheap junk...... far beter off with a Linksys SPA-2002 |
| 17:40.34 | JohnRobert | anyway, i gotta go, dinner..back in a bit |
| 17:40.43 | FuriousGeorge | but i assume its the zap module because if i try to just run asterisk -c from an init script i get the same problem |
| 17:40.57 | FuriousGeorge | and the CLI says "no such device zap-5" |
| 17:41.33 | FuriousGeorge | but if i log in and run asterisk -c it works. until i try safe_asterisk. so right now im just running asterisk in screen on this one server |
| 17:42.08 | stoffell | i have a problem with calls to international cell phone, they sometimes give chanunavail when not answered (http://pastebin.ca/164129), anyone ever experienced this? |
| 17:42.18 | FuriousGeorge | i dunno, maybe ill try the gentoo init scripts again, but those things almost never work for asterisk |
| 17:42.40 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 17:43.04 | [TK]D-Fender | FuriousGeorge: Pastebin everything surrounding that error message |
| 17:44.31 | FuriousGeorge | [TK]D-Fender: there is no error message other than "asterisk died" nothing gets written to car/log/asterisk/messages. i dont want to stop the service now b/c its business hours, but if i run asterisk -c from an init script (as opposed to safe asterisk), i see the same exact message as when i try to start asterisk without the ectdm modules loaded |
| 17:44.35 | FuriousGeorge | "no such device" |
| 17:44.45 | joako | anyone got a UTstarcom WiFI phone to register with asterisk? |
| 17:44.53 | FuriousGeorge | *wctdm |
| 17:45.14 | [TK]D-Fender | FuriousGeorge: Hrm.... could be an init problem... |
| 17:45.44 | syzygyBSD | [TK]D-Fender: ya, I would have thought so too, but it is dying on this server, debugging other peoples machines can be hard |
| 17:45.45 | FuriousGeorge | [TK]D-Fender: as in with my scripts? my gentoo scripts never work so on all my other servers i just replace the start script with "safe_asterisk" |
| 17:46.04 | syzygyBSD | I think it has something to do with them running a 64 bit smp kernel on a single proc machine though |
| 17:46.08 | FuriousGeorge | but i mean that i cant even run safe_asterisk from the command line |
| 17:46.14 | FuriousGeorge | all scripts aside |
| 17:46.50 | fafnir | anyone know which voip provider shido6 runs/works for? |
| 17:47.20 | kph100 | i need to create extensions context while using realtime. Is this possible? |
| 17:47.43 | *** join/#asterisk Qb3rt (n=jhgjkgui@58.68.252.216.dsl1.colba.net) |
| 17:51.07 | jhiver | Sep 11 19:48:03 ERROR[7743]: src/chan_h323.c:2476 load_module: Capabilities failure for OOH323. OOH323 Disabled. |
| 17:51.12 | jhiver | any idea what that means? |
| 17:51.35 | jhiver | i have compiled asterisk 1.2 + addons chan_ooh323 and i get this when i start it up |
| 17:51.35 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 17:52.16 | ramtha | sounds like unstable h323 src and oh323 is disabled by default? |
| 17:53.44 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:57.02 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-166-30.red.bezeqint.net) |
| 17:59.00 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 17:59.02 | jhiver | actually, it's 1.2 branch |
| 17:59.23 | jhiver | i have compiled asterisk, installed, and then same for asterisk-addons/asterisk-ooh323c |
| 18:01.54 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:03.47 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 18:03.50 | joako | Anyone ever work with the UTStarcom phones? I got an F3000 but it does not send REGISTER messages..... |
| 18:04.37 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 18:05.55 | *** join/#asterisk oej (n=oej@64.251.112.98) |
| 18:06.16 | *** join/#asterisk victim (n=none@ppp-71-136-55-174.dsl.sndg02.pacbell.net) |
| 18:08.45 | puzzled | joako: I had to upgrade an F1000 and F1000G to the latest firmware before it did anything |
| 18:11.19 | joako | Where can I D/l it? |
| 18:11.29 | *** join/#asterisk anthonyl (n=anthonyl@c-67-167-214-149.hsd1.il.comcast.net) |
| 18:11.49 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-e209ce21fca03c89) |
| 18:11.50 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 18:12.01 | joako | voipsupply sucks... whenever you request fimware they take a week to send it... and then I even want to pay them for the $90 support option but they refuse to do anything but email you back.... |
| 18:12.19 | JohnRobert | right |
| 18:12.27 | JohnRobert | maybe it's just the iax support that's not working.. |
| 18:16.46 | afrosheen | voipsupply has always been good to us |
| 18:17.00 | afrosheen | but we mainly deal with polycoms so firmware is easy to find |
| 18:17.29 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:17.38 | puzzled | joako: iirc sipgate has it. google around |
| 18:19.27 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 18:19.48 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 18:20.54 | Jason99 | I have a server with a Sangoma card and context that forwards all inbound IP calls to the Sangoma and all inbound TDM calls to another asterisk server. DTMF is working properly. What should the dtmfmode be in the sip.conf? |
| 18:21.46 | joako | Puzzled: sipgate.co.uk? I saw their site already, it only has the manuals |
| 18:21.52 | [TK]D-Fender | Jason99: RFC2833 wherever possible |
| 18:22.29 | puzzled | joako: well, I found the firmware for the F1000 and F1000G on their site. maybe look again |
| 18:23.03 | Jason99 | [TK]D-Fender: so for inbound TDM calls, I assume Asterisk does the DTMF detection and sends it in the RTP as rfc2833? |
| 18:23.20 | flujan | guys is possible to send a reload command using the asterisk manager? |
| 18:23.57 | afrosheen | flujan: probably |
| 18:24.17 | joako | Yep... on the same FAQs page, they have the F1000 but not F3000 |
| 18:24.23 | afrosheen | flujan: I know freepbx does that everytime you change an extension or anything else |
| 18:25.14 | flujan | afrosheen, hum... but i am using asterisk. :D |
| 18:25.57 | afrosheen | flujan: yeah I know, freepbx is a browser-based gui tool for asterisk..so if it can reload via the asterisk manager, there is a way |
| 18:25.59 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 18:26.11 | CunningPike | I have a call path like this: Asterisk1 -> IAX -> Asterisk2 -> IAXy. If the call doesn't go through on the IAXy, I would like to go to VoiceMail() on Asterisk1 - what do I need to do? |
| 18:26.16 | *** join/#asterisk file (n=file@neutrino.joshua-colp.com) |
| 18:26.16 | *** mode/#asterisk [+o file] by ChanServ |
| 18:27.27 | smackus | looking for some troubleshooting assistance. when i have a call into one of my extensions, and I transfer it to another extension... the call goes through, and it shows on the display as still being connected, but we cannot hear each other. if i dial the full local telephone number of that phone when I transfer everything works like it should. it only does this when i transfer to an internal extension. what could be going on? |
| 18:27.56 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 18:28.01 | CunningPike | smackus: How are you transferring? And are you using reinvite? |
| 18:28.24 | smackus | reinvite in the sip.conf? reinvite=yes? |
| 18:28.40 | joako | Cunning: if the call fails on Asterisk2 going to the IAXy the call from Asterisk 1 to Asterisk 2 would not be successful, so you would setup voicemail on #1 as if the iaxy was conected diretly... |
| 18:29.19 | smackus | it does not look like it. on my softphones, i have canreinvite=no and nothing on the polycom entries |
| 18:29.23 | *** join/#asterisk Skarmeth (n=Skarmeth@201009100056.user.veloxzone.com.br) |
| 18:29.58 | smackus | so should i have canreinvite=yes on my polycom entries? |
| 18:30.19 | afrosheen | smackus: I wouldn't do that if I were you |
| 18:30.20 | [TK]D-Fender | smackus: "canreinvite=no" in [general]. Period. |
| 18:30.34 | afrosheen | polycoms and reinvite are enemies |
| 18:31.06 | [TK]D-Fender | Reinvites ANWHERE are enemies. |
| 18:31.25 | CunningPike | joako: That's what I thought, but the call just hangs up. I have Dial(IAX2/Asterisk2/exten,20,o) on Asterisk1, and Dial(IAX2/iaxy) on Asterisk2 |
| 18:31.49 | smackus | CunningPike: was that not what you were asking me? if i have do not have reinvite=no what does it default to? |
| 18:31.58 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 18:32.08 | CunningPike | joako: I have tried Dial(IAX2/iaxy,20,o) on Asterisk2 followed by Congestion to no avail. Also tried Busy |
| 18:32.57 | CunningPike | smackus: You should have reinvite=no |
| 18:33.06 | afrosheen | smackus: the default should be 'no' |
| 18:33.10 | smackus | ok. would that affect my transfers? |
| 18:33.31 | Cresl1n | Hey all, just in case anybody cares, I got an ANSI style SS7 link to terminate calls with libss7 this weekend |
| 18:33.34 | syzygyBSD | any idea why when I try to transfer an IAX extension between 2 sip extensions it is failing with "500 "Internal Server Error" back from 192.168.0.182" (Phone's IP) |
| 18:33.36 | afrosheen | it affects everything..reinvite allows phones to bridge calls and take asterisk out of the loop, and that hurts |
| 18:33.38 | *** join/#asterisk kindor (n=roy@office.open-ict.nl) |
| 18:33.39 | kindor | hi |
| 18:33.39 | *** join/#asterisk AnthC (n=aclark@168.158.193.130) |
| 18:33.42 | kindor | anyone know if the Free Busy Light panel on the siemens optipoint 410s series work with asterisk? |
| 18:33.50 | Cresl1n | so all of you here in #asterisk with ANSI SS7 links and have been chompin at the bit to use it in asterisk, now's your chance :-) |
| 18:34.34 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:35.00 | smackus | but doesnt the peer type also specify the reinvite status? so that would have already been set to no if i understand correctly. So that should not be the cause of my transferring issues |
| 18:35.59 | afrosheen | smackus: I don't think the peer type defines that explicitly, I could be wrong |
| 18:37.35 | joako | CurringPike: so in extensions.conf you have exten => somext,1,dial(IAX2/asterisk2/iaxy) exten => somext,2,Congestion??? |
| 18:38.02 | niter3 | Hey guys, when dialing out on one of my trunks after pushing 9 I would like to continue to hear the dial tone before proceeding with the phone number. I guess you would call it a comfort tone. Is there a way I can acheive this? |
| 18:38.53 | momelod | anyone know where i can lookup milliwatt numbers? |
| 18:39.05 | momelod | my local provider (rogers) wont share :( |
| 18:39.23 | AnthC | convertit |
| 18:39.25 | AnthC | .com |
| 18:40.35 | joako | CunningPike: if anything try Dial(IAX2/Asterisk2/exten,20,g).. but if the call on asterisk2 is never answered it shouldnt be needed.... but try to avoid to answer ringing calls..... |
| 18:40.43 | momelod | niter3: ignorepat => 9 |
| 18:41.02 | RoyK | wtf is milliwatt numbers? |
| 18:41.12 | *** join/#asterisk Magicianx (n=magician@116-22.dr.cgocable.ca) |
| 18:41.18 | niter3 | momelod: just add that anywhere in the context? |
| 18:41.22 | JohnRobert | stupid atcom POS |
| 18:41.24 | momelod | its the phone companys number which is used for testing |
| 18:41.29 | JohnRobert | just doesn't work. |
| 18:41.41 | momelod | niter3: to the relevent context.. |
| 18:42.18 | niter3 | but anywhere in that context |
| 18:42.38 | momelod | i dunno google ignorepat? |
| 18:42.50 | CunningPike | joako: On Asterisk1, I have Dial(IAX2/Asterisk2/iaxy,20,o), followed by the usual Voicemail lines, but it never gets there - just hangs up |
| 18:42.52 | *** join/#asterisk dalekurt (n=dalekurt@65.183.3.229) |
| 18:44.13 | Jason99 | I have an AudioCodes Mediant 2000 connecting to an asterisk server and when I dial digits from the T1 connected to the Mediant, the Asterisk debug says "rtp.c:576 ast_rtp_read: Unknown RTP codec 96 received" |
| 18:44.40 | file | switch it to use payload 101 |
| 18:44.41 | afrosheen | niter3: are you actually having people dial 9 as a prefix just for the hell of it or does a PBX in the loop need it to let you dial out? |
| 18:44.51 | Jason99 | file: Let me try |
| 18:45.17 | niter3 | afrosheen: people have to push 9 before dialing out |
| 18:45.18 | Jason99 | file: you mean the Mediant correct? |
| 18:45.22 | file | yes |
| 18:45.35 | JohnRobert | does anyone here have an atcom AG-168V? |
| 18:45.36 | niter3 | after pushing 9 I would like them to still hear a dial-tone |
| 18:45.52 | afrosheen | niter3: so you want it to mimic your oldschool pbx, I see |
| 18:46.06 | niter3 | pretty much |
| 18:46.10 | niter3 | if that's what you would call it |
| 18:46.12 | afrosheen | we had to train the new hires here to stop hitting 9...we don't use that here |
| 18:46.38 | CunningPike | joako: The 'g' option did it - thanks! |
| 18:47.06 | [TK]D-Fender | niter3: What kind of phones are you using? |
| 18:47.39 | *** join/#asterisk Adrian__ (n=foo@zux221-091-123.adsl.green.ch) |
| 18:47.44 | [TK]D-Fender | ZI have banished Dial-9 from all the system I deploy. its a decrit idea that should be allowed to die |
| 18:47.47 | Jason99 | file: I changed the Mediant to 101 and it still said the same message |
| 18:48.26 | file | Jason99: do an rtp debug, see where payload 96 is coming from - and set it to be 101, if you did and it's still 96 - then it didn't take effect probably |
| 18:48.41 | Jason99 | file: ok thanks.. |
| 18:49.03 | sivana | does anyone know anything about video conferencing and asterisk |
| 18:49.10 | momelod | when im tuning my rx and tx gains, where should i aim for? some say the ztmonitor should show the rxgain maxed out, while others say 50%? |
| 18:49.58 | afrosheen | momelod: 75% |
| 18:50.02 | afrosheen | happy medium |
| 18:50.16 | *** join/#asterisk umay (n=chris@71-208-192-243.hlrn.qwest.net) |
| 18:50.25 | momelod | is that based on anything or are you just amusing yourself? |
| 18:50.47 | afrosheen | it's what I did before I did an atomic elbow drop to our zaptel card |
| 18:50.53 | Jason99 | file: I was changing it at the wrong place, changed it to 101 and that worked perfectly.. thanks a lot |
| 18:50.58 | momelod | lol |
| 18:50.59 | afrosheen | we had 2 of the modules go bad on it |
| 18:51.10 | *** join/#asterisk hypa7ia (i=hypatia@judecca.aculei.net) |
| 18:51.25 | afrosheen | thank god we're 100% sip now |
| 18:51.48 | dalekurt | can I get some help, I'm getting Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface when I run ztcfg |
| 18:51.59 | dalekurt | sorry, Unable to open master device '/dev/zap/ctl' |
| 18:52.25 | dalekurt | I'm told it has something to do with udev |
| 18:53.15 | afrosheen | dalekurt: did you read the readme.linux26 and readme.udev that came with the zap source? |
| 18:53.52 | dalekurt | can't say i did |
| 18:53.55 | dalekurt | ;( |
| 18:55.05 | JohnRobert | what would 'prack' mean on an ata control panel thing? |
| 18:55.13 | JohnRobert | the manual for my ata is very poor. |
| 18:55.17 | afrosheen | dalekurt: there's your starting point :) |
| 18:57.28 | niter3 | [TK]D-Fender: A simple linksys adapter to a standard phone right now |
| 18:58.10 | dalekurt | thanx. |
| 18:59.02 | *** join/#asterisk fafnir_ (n=notfaf@unaffiliated/fafnir) |
| 18:59.27 | [TK]D-Fender | niter3: Unless they added the equivalent of "ignorepat" to their own dialplan syntax the answer is "not possible" then. |
| 18:59.31 | joako | Anyone ever get a UTStarcom phone to register with ASterisk? |
| 19:00.01 | JohnRobert | I need to return this atcom ata's :/ |
| 19:00.09 | JohnRobert | they're blatantly broken or something. |
| 19:00.10 | niter3 | [TK]D-Fender: what do you mean by that? If they added it to the ATA linksys adapter? |
| 19:00.20 | JohnRobert | well, actually it looks like they were never programmed to work |
| 19:00.52 | [TK]D-Fender | niter3: You wanted to hear dialtone after punching in the 9 prefix to an external call. That is something that would have to be done within the ATA, not *. |
| 19:01.04 | niter3 | oh ok.. |
| 19:01.10 | niter3 | thanks... |
| 19:01.23 | niter3 | But like mentioned earlier, this is old school style of doing things? |
| 19:01.35 | afrosheen | JohnRobert: the digium iaxy comes highly recommended |
| 19:02.22 | CunningPike | joako: Yup - our help desk uses one |
| 19:02.32 | joako | How do you get it to register? |
| 19:02.34 | sx-wks | afrosheen: they don't self destruct anymore ? |
| 19:02.46 | [TK]D-Fender | niter3: Yes, and completely worthless. You should get rid of prefixs like that |
| 19:02.46 | afrosheen | sx-wks: not after they changed them to the cobra style |
| 19:02.50 | joako | I setup sip.conf, setup the phone, I get no "Registered xxx" Or "Registration failed" messages |
| 19:02.51 | sx-wks | hah |
| 19:03.14 | niter3 | [TK]D-Fender: Probably will.. I just have to learn how to program prefixs in. I'm still very new to *. |
| 19:03.15 | joako | If I do sip debug I get some messages, but it never seems to try to register... |
| 19:03.18 | CunningPike | joako: Much the same as any other SIP UA - I don't have it with me at the moment, but afaik there are a bunch of menu options on the phone |
| 19:03.45 | sx-wks | afrosheen: what's the diff between the "cobra" style and whatever style that was before ? |
| 19:03.46 | CunningPike | joako: Oh, and my problem was caused by an Answer() in a macro I was running...... |
| 19:04.13 | CunningPike | joako: If you can wait until tomorrow, I can get the phone and pm you through it? |
| 19:04.27 | afrosheen | sx-wks: slightly different hardware, the cobra ones are the ones they build now, hit digium's site and you'll see what I mena |
| 19:05.12 | afrosheen | http://www.digium.com/en/products/hardware/s101i.php |
| 19:06.44 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 19:08.58 | sx-wks | afrosheen: ah, the totally gray instead of bi-color case :D I see |
| 19:10.49 | joako | Cunning: pm me THROUGH the phone? |
| 19:16.40 | *** join/#asterisk foxmjay (i=root@adsl196-172-99-217-196.adsl196-12.iam.net.ma) |
| 19:16.49 | CunningPike | joako: Nah - pm here - walk you through setting it up |
| 19:17.23 | CunningPike | Lunchtime |
| 19:18.01 | *** join/#asterisk _fa_ (i=faceoff@devel4.net) |
| 19:18.02 | fafnir_ | hi, anyone here have a softfone, and zfone, want to help me out with something? |
| 19:19.15 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 19:22.44 | *** join/#asterisk _fa_ (n=faceoff@devel4.net) |
| 19:22.45 | _fa_ | Hi |
| 19:23.00 | _fa_ | Do sb know how to play a voice message in call background, some notification about prepaid account balance or sth. |
| 19:23.19 | bkw_ | show application dial |
| 19:23.22 | bkw_ | it has the L option |
| 19:23.25 | bkw_ | does exactly what you want |
| 19:23.29 | _fa_ | thank you |
| 19:26.10 | _fa_ | bkw_ can you tell me in which files should I look if I want to customize more the L option? |
| 19:26.24 | bkw_ | look on voip-info |
| 19:26.55 | bkw_ | its done from the dialplan |
| 19:26.55 | _fa_ | I know |
| 19:26.55 | _fa_ | but there is only three option, and this is not enought for me, I want to modify that in source code |
| 19:27.02 | justinu|laptop | app_dial.c |
| 19:27.19 | _fa_ | Hm, ok, thanks |
| 19:27.38 | RoyK | bkw_: do they allow you in here?? |
| 19:28.00 | _fa_ | heh ;] I allow |
| 19:28.56 | jamincollins | is there any way in a TDM <> * <> SIP environment to know why Asterisk decided to take down both the TDM and SIP legs of a call? |
| 19:30.31 | RoyK | jamincollins: i guess that's called debugging |
| 19:30.58 | jamincollins | both PRI and SIP debugging only show that * generated the disconnect, nothing about /why/ |
| 19:31.37 | jamincollins | seemingly out of the blue, * generates a disconnect on the PRI followed by a SIP BYE message |
| 19:31.53 | RoyK | 1.2 or trunk? |
| 19:31.59 | jamincollins | 1.2.10 |
| 19:32.27 | RoyK | 1.2.12 does the same also? |
| 19:32.41 | RoyK | you know, there might be fixes between those two :P |
| 19:32.50 | jamincollins | using the Debian packages, don't have a 1.2.12 package to try at this time |
| 19:33.00 | RoyK | use the source, luke |
| 19:33.21 | RoyK | jamincollins: you will not get any support or even positive feedback if not using the latest version |
| 19:33.24 | foxmjay | i got chan_phone.c:41:29: linux/compiler.h : file doesn't exist , while compiling asterisk 1.2.12 with the kernel-2.6.17 , what's wrrong ? plz |
| 19:34.08 | jamincollins | RoyK: I'm just asking for debugging guidance... is there a way to find /why/ * triggered the disconnect? |
| 19:35.32 | niter3 | hey guys how can i have a context that allows me to just dial instead of pushing 9. exten => _9.,1,Dial(......... |
| 19:35.44 | niter3 | do I just do a exten => _.,1,Dial |
| 19:35.44 | niter3 | ? |
| 19:36.58 | [TK]D-Fender | niter3: I'd suggest doing multiple dials to account for valid patterns. _X11 for services, _XXXXX. as a catch-all for anything else perhaps |
| 19:37.42 | [TK]D-Fender | niter3: And ensure that your internal purpose numbers are 2 or 4 digits for example as to avoice clashing with X11 |
| 19:38.26 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 19:38.53 | *** join/#asterisk pdt (n=ptinsley@adsl-154-211-201.ard.bellsouth.net) |
| 19:40.41 | evildeshi | anyone know what would cause audio to not get sent at all weather i use meet me or direct calling of other people no audio is passing through either way |
| 19:41.06 | *** join/#asterisk tpak (n=tpak@69-162-129-127.clspco.adelphia.net) |
| 19:41.39 | *** join/#asterisk jmls (n=asterisk@host81-159-195-120.range81-159.btcentralplus.com) |
| 19:41.55 | niter3 | [TK]D-Fender: yah you lost me.. |
| 19:42.05 | niter3 | What are these called _9 - dialing prefixes |
| 19:42.06 | _fa_ | RoyK do you know si it possible in N option not to end call, but only jump to some priority in some context? |
| 19:42.13 | niter3 | If I can I will just pull it up on google. |
| 19:42.20 | *** part/#asterisk vosque (i=e40djoee@69.50.222.162) |
| 19:42.24 | _fa_ | Or maybe N generate n+1 priority? |
| 19:42.39 | *** join/#asterisk EvilDeshi (n=evildesh@pool-71-98-10-135.mdsnwi.dsl-w.verizon.net) |
| 19:42.39 | bkw_ | yes |
| 19:43.19 | _fa_ | So How Can I determine is it effect of N function or calleed just hangup? |
| 19:44.55 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.42.141) |
| 19:45.57 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 19:46.57 | smackus | i am playing with call-limit in sip.conf... I really like what it does, but is there a way to make it for incoming call only? so i limited to the number of incoming calls, but not outgoing calls? |
| 19:48.10 | *** join/#asterisk fafnir_ (n=notfaf@unaffiliated/fafnir) |
| 19:48.50 | *** part/#asterisk ToxaP (n=ToxaP@213.227.193.75) |
| 19:49.01 | smackus | or better yet, i am trying to figure out how to use hint |
| 19:49.26 | smackus | i have my hit priority and in sip.conf i have notifyringing = yes |
| 19:50.07 | smackus | how do i make it so that it just gives a busy signal back instead of presenting the call? |
| 19:52.17 | *** join/#asterisk ppdm (i=pdm@pdm.sh) |
| 19:52.24 | ppdm | good afternoon :) |
| 19:52.28 | momelod | anyone know the milliwatt test line for rogers (in toronto, ontario, canada) |
| 19:52.30 | ppdm | anyone have any experience/luck with 7961? |
| 19:52.42 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-e209ce21fca03c89) |
| 19:53.05 | eliXier | smackus exten => 1002,hint,SIP/1002 (1002 = user/line) |
| 19:53.08 | eliXier | thats it |
| 19:53.19 | eliXier | in extensions.conf |
| 19:53.31 | eliXier | [default] |
| 19:54.31 | smackus | so nothing has to be done in sip.conf? |
| 19:55.18 | ramtha | no |
| 19:55.32 | [TK]D-Fender | smackus: :Check the status in your dialplan before even calling that device |
| 19:55.40 | smackus | yeah, its line one |
| 19:55.57 | [TK]D-Fender | smackus: Whats line 1? |
| 19:56.08 | smackus | exten => s,hint,SIP/test_${EXTEN} |
| 19:56.36 | [TK]D-Fender | smackus: That doesn't work at all. You can evaluate anything in a hint and that has nothing to do with limiting calls to a device. |
| 19:56.55 | [TK]D-Fender | smackus: That line is completely wrong. |
| 19:57.19 | smackus | ok, i was just using the example from the wiki |
| 19:57.40 | smackus | Example exten => 200,hint,SIP/phone1 ; this is case sensitive (!) in 1.0.9 and 1.2.0 |
| 19:57.45 | smackus | am i misunderstanding? |
| 19:57.47 | [TK]D-Fender | smackus: Even if it could evaluate its come out as exten => s,hint,SIP/test_s (because exten=s) |
| 19:58.01 | [TK]D-Fender | smackus: Hints will NOT help you limit calls. Clear? |
| 19:58.25 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
| 19:58.28 | kink0 | hello |
| 19:58.34 | smackus | yeah... i changed gears... all i want is to playback a busy signal if this extension is on the phone. |
| 19:58.55 | [TK]D-Fender | smackus: Then before issuing your dial you should check to see if they're on a call. |
| 19:59.01 | kink0 | I still getting " channel.c:787 channel_find_locked: Avoided deadlock for '0x8332258'" but SIP calls are passed ok |
| 19:59.06 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 19:59.08 | tpak | anyone on that is part of the rpath #key project? |
| 19:59.09 | smackus | doesnt hint do that? |
| 19:59.13 | kink0 | and h323 are completelly ignored |
| 19:59.31 | kink0 | anyway to reset just h323 without loss CDR ? |
| 19:59.42 | bkw_ | resetCDR(w) |
| 20:00.09 | [TK]D-Fender | smackus: Perhaps I haven't made this clear enough. *NO* |
| 20:00.42 | bkw_ | kink0: let me translate that for you... *PUNT* |
| 20:00.46 | [TK]D-Fender | smackus: Hint is a priority so that one phone can SEE the status of a given device. it in no way AFFECTS its ability to dial it however. |
| 20:01.08 | drfreeze | Dial(Zap/1-1/${EXTEN}) - dials out on group 1 line 1 |
| 20:01.15 | drfreeze | what is group 1 line 2? 1-2 or 2-1 ?? |
| 20:01.32 | [TK]D-Fender | drfreeze: Dial(Zap/1/${EXTEN}) |
| 20:01.56 | kink0 | bkw_ *PUNT* ? |
| 20:02.01 | drfreeze | D-Fender and Dial(Zap/2/${EXTEN}) for line 2? |
| 20:02.02 | smackus | [TK]D-Fender: right, but based off of what the hint reports, cant i have another priority that tells it to do something? |
| 20:02.05 | [TK]D-Fender | drfreeze: 1-1 is a channel designation, not a device designation. |
| 20:02.31 | drfreeze | I thought it was channel and device designation |
| 20:02.32 | drfreeze | no? |
| 20:02.38 | [TK]D-Fender | smackus: No. Hint is only for displaying status to a device POLLING for it. the is no action related to it. |
| 20:02.48 | [TK]D-Fender | drfreeze: No. |
| 20:03.17 | [TK]D-Fender | drfreeze: You either specify a device or a group. |
| 20:03.21 | smackus | I guess i am just not getting my head around this. there has to be some easier way to make it just give a busy signal that the ways I have been shown. |
| 20:03.24 | drfreeze | What is channel 2? |
| 20:03.28 | drfreeze | 2-2? |
| 20:04.52 | afrosheen | ok back |
| 20:05.05 | [TK]D-Fender | drfreeze: Best to forget about that.... its a hold-over..... |
| 20:05.30 | [TK]D-Fender | smackus: "show application chanisavail" <------ |
| 20:06.00 | drfreeze | [TK]D-Fender: what? the x-y notation? |
| 20:06.03 | [TK]D-Fender | smackus: YOU make it do what you want using the dialplan, and no there isn't a 1-stop-shop miracle line for that. |
| 20:06.25 | [TK]D-Fender | drfreeze: Yes, you don't ever refer to them like that in the dial-plan. |
| 20:06.47 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 20:07.03 | drfreeze | ugh. Ok. so I should use Zap/g1 for group 1 and Zap/1 for channel 1 of group?? |
| 20:07.36 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 20:07.39 | kink0 | the way to force hangup while deadlock , to avoid lose CDR was block udp transit, so the channel goes to hangs , then I need to restart Asterisk |
| 20:07.42 | [TK]D-Fender | drfreeze: First part is right, the second isn't. Zap/1 is a specific channel. Not as part of any group by its definition. |
| 20:07.52 | jamincollins | does noone know a means of locating what triggered * to tear down both the PRI and SIP legs of a call passing through it? |
| 20:07.55 | kink0 | but any other idea about my problem with deadlock and a lot of asterisk threads ? |
| 20:08.24 | mishehu | there once was a server who blocked a thread... |
| 20:09.15 | sivana | Is there any way to log remote unix connections in *? |
| 20:09.15 | drfreeze | [TK]D-Fender: I've got fxsks=1-4 in zaptel.conf. I have 4 incoming lines on a tdm card. So I reference these lines with Zap/x where x is 1-4? |
| 20:09.27 | afrosheen | sivana: um the box logs that itself I believe |
| 20:09.43 | kink0 | but there anyway to tell asterisk maximum threads to use ? why Asterisk starts new threads while deadlock ? |
| 20:09.47 | sivana | afrosheen: well, I'm getting remote sip reloads with NO logs |
| 20:09.53 | afrosheen | drfreeze: it's smarter to just use group zero or group 1, put all those channels together and letting * handle them |
| 20:10.06 | afrosheen | sivana: REMOTE sip reloads? |
| 20:10.07 | sivana | afrosheen: and I need to find who/what is doing it |
| 20:10.12 | sivana | yes |
| 20:10.20 | afrosheen | that's very strange, I didn't know that was possible even |
| 20:10.35 | sivana | I was watching console when this came across |
| 20:10.38 | sivana | -- Remote UNIX connection |
| 20:10.38 | sivana | <PROTECTED> |
| 20:10.44 | sivana | -- Remote UNIX connection disconnected |
| 20:10.47 | [TK]D-Fender | drfreeze: Yes, if you want to pick a SPECIFIC line to call out on. |
| 20:10.52 | drfreeze | afrosheen: I need to make a call on a specific line in this case. I.e, turning off call forwarding on individual lines |
| 20:11.06 | afrosheen | sivana: so your asterisk manager must be open and set to default..? |
| 20:11.14 | sivana | enabled=no |
| 20:11.26 | sivana | very strange happening |
| 20:11.28 | *** join/#asterisk joako (n=kvirc@64.238.175.93) |
| 20:11.32 | afrosheen | sivana: I'd firewall it and poke holes for * only |
| 20:11.40 | afrosheen | then see what happens |
| 20:11.55 | joako | Anyone ever get a UTStarcom phone to register with Asterisk? |
| 20:11.56 | drfreeze | [TK]D-Fender: I'm running 1.2.6. I still see things like Zap/47-1 on the wiki. Did this change happen recently? |
| 20:12.28 | [TK]D-Fender | drfreeze: You don't put it like that in the DIALPLAN. that can be cli output of a call in progress however. |
| 20:13.19 | drfreeze | [TK]D-Fender: ok, I see. |
| 20:13.20 | drfreeze | thanks |
| 20:13.21 | tzanger | ok |
| 20:13.25 | tzanger | I've got someone fucking around on my box |
| 20:13.30 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:13.37 | [TK]D-Fender | drfreeze: if SIP/phone1 is on a call the CHANNEL will be called something like SIP/phone1-a54d . The "-a54d" suffix is to make that specific call "unique" |
| 20:13.44 | tzanger | ssh shows me logged in but netstat shows another ssh session, which is using pts/2 and sitting at a bash script |
| 20:13.47 | tzanger | er shell prompt |
| 20:14.00 | [TK]D-Fender | tzanger: Make sure they are using condoms... God only knows where your box has been! |
| 20:14.01 | tzanger | I can peer into the shit on /proc/[pid] but is there a way to watch stdin/out |
| 20:14.08 | tzanger | [TK]D-Fender: yeah I'm kind of slutty that way |
| 20:14.17 | Zodiacal | anyone kwow what _ does in front of an ext.? i can't find doc's that explain that.. |
| 20:14.50 | [TK]D-Fender | Zodiacal: Pattern matching.... this is diaplan 101..... |
| 20:14.52 | tzanger | I wonder if I can write a quick program to duplicate /dev/pts2... |
| 20:15.13 | Zodiacal | tkd-fender i thought X and .'s etc.. did that |
| 20:15.20 | *** join/#asterisk markQing (n=mark@82-197-202-242.dsl.cambrium.nl) |
| 20:15.32 | afrosheen | tzanger: can you redirect stdout from pts2 to your current shell? |
| 20:15.37 | *** part/#asterisk tpak (n=tpak@69-162-129-127.clspco.adelphia.net) |
| 20:15.44 | tzanger | afrosheen: that's what I'm trying to do |
| 20:16.09 | [TK]D-Fender | Zodiacal: "_" tells * to LOOK at the X's, and so on... |
| 20:16.12 | *** join/#asterisk beu (i=beu@freenode/developer/gentoo.developer.beu) |
| 20:16.30 | Zodiacal | tkd-fender ahh, so no wildcards with out _? |
| 20:16.40 | *** join/#asterisk niZon (n=bleh@S0106beefd4cecc3d.wp.shawcable.net) |
| 20:16.55 | [TK]D-Fender | Zodiacal: exten => _5XX,1,SayNumber(${EXTEN}) |
| 20:17.14 | [TK]D-Fender | Zodiacal: Correct, you will be sitting around wondering why you can't dial anything... |
| 20:17.18 | Zodiacal | tkd-fender Thank You! |
| 20:17.49 | *** join/#asterisk file2 (n=IrcNet@out.clearnet.com) |
| 20:17.49 | *** mode/#asterisk [+o file2] by ChanServ |
| 20:18.52 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 20:21.23 | EvilDeshi | ok im using asterisk @home 2.8 and asterisk is running but when i issue asterisk -r i get Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 20:22.03 | jamincollins | RoyK: once I upgrade to 1.2.12, if the problem is still present, do you have any suggestions on how to locate what is triggering the disconnect? |
| 20:22.11 | joako | is it running already? |
| 20:23.06 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 20:23.19 | [TK]D-Fender | EvilDeshi: What tells you * is running? |
| 20:25.21 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 20:26.24 | *** join/#asterisk beu (i=beu@freenode/developer/gentoo.developer.beu) |
| 20:26.27 | *** join/#asterisk sm7syx (n=testuser@h-83-140-228-76.ip.rixbredband.se) |
| 20:27.17 | *** join/#asterisk dmz (n=dmz@64.20.150.9) |
| 20:27.22 | dmz | howdy y'all |
| 20:27.41 | file2 | hi |
| 20:27.42 | dmz | can anyone recommend an IAX voip provider for a small business (ie 2-5 lines with rollover)? |
| 20:28.10 | sm7syx | Hi, my asterisk is not listening on port 5060 ! I have tried to find aswers on the web but most of them is about IPTABLES and my firewall is DOWN ! Other suggestions? |
| 20:28.13 | ramtha | dmz: where located? |
| 20:28.19 | dmz | US |
| 20:28.26 | ramtha | sorry |
| 20:28.34 | dmz | :( |
| 20:28.42 | dmz | brb daughter waking up from nap |
| 20:29.51 | EvilDeshi | [TK]D-Fender ps -aux shows asterisk running |
| 20:30.05 | EvilDeshi | asterisk 4010 0.0 3.0 19172 7712 ? Sl 15:19 0:00 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c |
| 20:30.11 | joako | DMZ: I could RECOMMEND one but they dont have the cheapest prices for DIDs |
| 20:30.52 | [TK]D-Fender | EvilDeshi: Could be a permissions issue to the PID or that its simply not where expected. |
| 20:31.31 | EvilDeshi | i dunno how to fix anything on this asterisk@home deal this was a fresh default install as of an hour ago |
| 20:31.45 | EvilDeshi | if i kill asterisk i just restarts |
| 20:31.48 | EvilDeshi | it |
| 20:32.07 | *** join/#asterisk sevard (n=sev@c-67-188-173-23.hsd1.ca.comcast.net) |
| 20:32.24 | C6Vette | EvilDeshi, Kill safe_asterisk, that will restart asterisk when you kill it |
| 20:32.51 | sevard | more importantly, killall -9 safe_asterisk |
| 20:33.07 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 20:33.26 | EvilDeshi | oh that killed it finally |
| 20:33.36 | EvilDeshi | but now when i load up with asterisk -vvvvvvv i get Sep 11 15:33:01 WARNING[4189]: loader.c:325 __load_resource: libspandsp.so.0: cannot open shared object file: No such file or directory |
| 20:33.37 | EvilDeshi | Sep 11 15:33:01 WARNING[4189]: loader.c:554 load_modules: Loading module app_rxfax.so failed! |
| 20:33.37 | EvilDeshi | Ouch ... error while writing audio data: : Broken pipe |
| 20:34.10 | C6Vette | take app_rxfax.so out of modules.conf |
| 20:34.45 | C6Vette | could have an error in a couple conf files. |
| 20:35.09 | EvilDeshi | haha thanks man |
| 20:35.14 | EvilDeshi | now i can get in the cli |
| 20:35.15 | *** join/#asterisk Katty (n=Katty@64.82.232.54) |
| 20:35.42 | file2 | kattttty |
| 20:35.52 | Katty | hihi |
| 20:36.22 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:36.24 | file2 | how ae chu' |
| 20:37.47 | *** join/#asterisk Skarmeth (n=Skarmeth@201009082012.user.veloxzone.com.br) |
| 20:38.41 | EvilDeshi | i cant seem to register my softfone using my external ip |
| 20:38.54 | EvilDeshi | no firewall no nat |
| 20:39.46 | C6Vette | EvilDeshi, What does it say on cli? |
| 20:39.48 | C6Vette | anything? |
| 20:39.51 | sm7syx | EvilDeshi, what does the cli say ? |
| 20:40.14 | EvilDeshi | nothing when i try to use external |
| 20:40.19 | EvilDeshi | works fine on the local ip |
| 20:40.32 | EvilDeshi | so i guess i haev nat but i disabled the firewall and set dmz to the asterisk box |
| 20:40.34 | sm7syx | EvilDeshi, but internaly it works ? |
| 20:41.09 | EvilDeshi | yeah |
| 20:41.55 | afrosheen | dmz = nat in most cases |
| 20:41.58 | sm7syx | Mine don't! Nothing happens.... Sorry that no help for you but I will follow the discussion |
| 20:42.14 | sevard | dmz is NOT nat |
| 20:42.23 | afrosheen | on our firewall and others, it is |
| 20:42.35 | Nugget | You are misusing the term "dmz" |
| 20:42.43 | sevard | dmz is demilitarized zone |
| 20:42.45 | sm7syx | Nugget, tnx |
| 20:42.55 | sevard | totally different than network address translation |
| 20:42.56 | afrosheen | yeah, I know what a dmz is, thanks |
| 20:43.12 | sevard | heh. |
| 20:43.13 | afrosheen | but firewalls such as the Sonicwall nat servers on the dmz |
| 20:43.40 | afrosheen | it isn't obvious from anywhere but the super detailed tech pdfs on it |
| 20:43.53 | Nugget | are you using the term "dmz" to refer to your public IP? If so, that's not what a DMZ is at all. |
| 20:44.21 | afrosheen | no, I'm not |
| 20:44.40 | *** join/#asterisk oej (n=oej@64.251.112.98) |
| 20:44.40 | afrosheen | EvilDeshi might though |
| 20:44.59 | Nugget | then I don't know what you mean by "dmz = nat in most cases" because that couldn't be farther from the truth. |
| 20:45.04 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-75020f7d3210c469) |
| 20:45.10 | Nugget | the majority of nat users have no dmz. |
| 20:45.14 | Nugget | the VAST majority |
| 20:45.51 | EvilDeshi | i am trying port forwarding now |
| 20:45.55 | afrosheen | I'm saying that dmz on firewalls such as the sonicwall introduces nat |
| 20:46.08 | afrosheen | it could be the case for him as well |
| 20:46.11 | CunningPike | Am I right in thinking that there is no way to obtain MWI from a server other than the one you're registered with? |
| 20:46.14 | sevard | dmz is a zone where all traffic on a router performing the functions of nat is forwarded to unless previously stated in routing rules |
| 20:46.30 | sevard | that is a non formal definition |
| 20:46.43 | sevard | but, case in point, dmz != nat, and you are stupid for saying so. |
| 20:46.48 | EvilDeshi | well now my phone registers |
| 20:46.51 | joako | CunningPike: it is possible, look for mwi + ser at voip-info.org |
| 20:46.52 | Nugget | the definition of dmz should not include nat at all. nat is not implied in any way. |
| 20:46.56 | EvilDeshi | now hoepfully that aduio issue i had earlier is gone |
| 20:47.07 | EvilDeshi | there was no audio coming or going except to external devices |
| 20:47.13 | CunningPike | joako: Ah yes - not running SER :) |
| 20:47.35 | joako | But it's a start... its possbile for asterisk to deliver the MWI other ways.... |
| 20:47.38 | Nugget | you can have (and many people do) a dmz without employing nat anywhere. |
| 20:47.45 | CunningPike | joako: OK - thanks |
| 20:47.53 | afrosheen | Nugget: correct, which is why we beat our heads against this sonicwall forever until we dual-homed it |
| 20:47.54 | EvilDeshi | nope same problem |
| 20:48.17 | EvilDeshi | if i call up the box using a regular phone and my ipkall number i can hear audio on teh phone |
| 20:48.21 | sevard | Nugget: of course, dmz applys to a firewall |
| 20:48.34 | EvilDeshi | but if i dial up with the soft fones or to another client extension i get no audio |
| 20:48.50 | afrosheen | EvilDeshi: are you forwarding the rtp ports like 10,000-20,000? |
| 20:48.59 | EvilDeshi | all ports 1 - 60000 |
| 20:49.08 | EvilDeshi | tcp and udp |
| 20:49.10 | joako | CunningPike: http://www.voip-info.org/wiki/view/Asterisk+at+large FIVE methods for mwi outside of asterisk..... |
| 20:49.26 | EvilDeshi | ive been reading about nat and sip issues |
| 20:49.38 | afrosheen | sounds like typical nat nonsense to me |
| 20:50.12 | afrosheen | EvilDeshi: is this at a company on a real connection or at your house |
| 20:50.15 | EvilDeshi | i wish strom was here |
| 20:50.46 | *** join/#asterisk daysmen3 (n=primus@host86-139-50-232.range86-139.btcentralplus.com) |
| 20:52.12 | dmz | joako, which one (voip provider) is that? |
| 20:53.31 | *** join/#asterisk ACiDV (n=acidvici@bas3-sherbrooke40-1177838783.dsl.bell.ca) |
| 20:54.44 | *** join/#asterisk afrosheen (n=test@txprotoa2.august.net) |
| 20:54.51 | ACiDV | Hi, anyone here use the Page command ? With asterisk 1.2.10 I have audio but on 1.2.11 and 1.2.12 I heard nothing and when I enable Full duplex, I get lots of error about Native format error. Page look broken on 1.2.11+ |
| 20:55.39 | *** join/#asterisk EvilDeshi (n=evildesh@pool-71-98-10-135.mdsnwi.dsl-w.verizon.net) |
| 20:57.50 | jamincollins | the zaptel-1.2.9 tarball from asterisk.org fails to build, apparently due to the tarball missing a required header... is there a workaround? |
| 20:58.36 | RoyK | jamincollins: not really, but try 1.2.12 first |
| 20:58.57 | jamincollins | RoyK: trying... zaptel won't compile... |
| 20:59.00 | Aurs | jamincollins: centos? |
| 20:59.10 | Aurs | spinlock.h |
| 20:59.12 | jamincollins | Aurs: no, not using CentOS |
| 20:59.17 | Aurs | typo in kernel sources in centos |
| 20:59.18 | Aurs | ok |
| 20:59.32 | jamincollins | zaptel-1.2.9/wct4xxp/../oct612x/include/oct6100api/oct6100_api.h |
| 21:00.04 | schotten83 | jamincollins, doesnt compile at all or just some packages? |
| 21:00.10 | sm7syx | what to do with "udp port 5060 unreachable" ? |
| 21:00.16 | jamincollins | the tarball does not contain the oct612x directory |
| 21:00.18 | schotten83 | can u paste @ pvt the compile error? |
| 21:00.23 | schotten83 | overhere thats normal |
| 21:00.58 | schotten83 | sm7syx certify that the 5060 udp port is not blocked at ur firewall |
| 21:01.05 | *** join/#asterisk dlynes_laptop (n=dlynes@s64-180-109-134.bc.hsia.telus.net) |
| 21:01.39 | schotten83 | jamincollins that at "make" ? |
| 21:01.45 | jamincollins | yes |
| 21:01.51 | sm7syx | firewall is DOWN ! Iptables is empty, first thing I checked. |
| 21:01.54 | dlynes_laptop | I'm curious |
| 21:02.12 | dlynes_laptop | How much memory should I typically have for asterisk on a 2 line, 6 extension system? |
| 21:02.27 | *** join/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
| 21:02.29 | afrosheen | dlynes_laptop: probably not much depending on codecs |
| 21:02.31 | dlynes_laptop | I'm guessing 256MB's should be more than sufficient, correct? |
| 21:02.51 | *** part/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
| 21:02.57 | afrosheen | our * box has 50 extensions and 4 trunks, on decent hardware, barely peaks at over 512M at any given timee |
| 21:02.57 | dlynes_laptop | afrosheen: Everything's gsm to the softswitch from the pbx |
| 21:03.04 | dlynes_laptop | afrosheen: and ulaw, internally |
| 21:03.19 | afrosheen | 256 is probably fine as long as your linux distro isn't bloaty |
| 21:03.29 | dlynes_laptop | afrosheen: far from it...i'm using slackware |
| 21:03.31 | schotten83 | sorry, Im withou access to my asterisk server now.. |
| 21:03.41 | afrosheen | then you're good to go |
| 21:03.45 | *** join/#asterisk adorah (n=admin@84.94.122.156.cable.012.net.il) |
| 21:03.47 | schotten83 | dlynes_laptop if the box is just for that it would be ok |
| 21:03.58 | afrosheen | * just isn't that greedy with ram |
| 21:04.13 | dlynes_laptop | afrosheen: it just seemed odd because I was maxing out at 256MB's before, so i doubled the memory to 512MB's, and it's 40MB's away from max now |
| 21:04.13 | schotten83 | hmm |
| 21:04.14 | jamincollins | schotten83: no worries... the problem has been disconnecting people for a week now... another day won't matter much |
| 21:04.35 | dlynes_laptop | And i've got 468MB's being used by the SIP channel driver |
| 21:04.55 | dlynes_laptop | And there's no conversations currently happening |
| 21:05.11 | dlynes_laptop | you'd think the memory allocation for the sip channel driver should be close to zero at the moment |
| 21:05.20 | *** join/#asterisk Defraz (n=t0tal@032-252-542.area1.spcsdns.net) |
| 21:05.29 | dlynes_laptop | and another 66MB's for m-o-h which isn't running right now, either |
| 21:05.34 | afrosheen | dlynes_laptop: don't forget that linux will use your ram, any ram not being used is wasted, just watch that swapping |
| 21:05.36 | RoyK | jamincollins: that's a known problem |
| 21:05.45 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 21:05.47 | JohnRobert | ram is cheap |
| 21:05.49 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:05.49 | JohnRobert | buy some more |
| 21:05.50 | JohnRobert | :p |
| 21:05.51 | RoyK | jamincollins: there's a bug in latest zaptel. check out from svn instead |
| 21:05.56 | dlynes_laptop | afrosheen: This is ram being used by asterisk, not linux |
| 21:06.04 | SomeJ | dlynes_laptop not sure if this is right or not, but iv been told in the past that linux will some times alocate ram and not free it till it is needed for something else |
| 21:06.04 | dlynes_laptop | afrosheen: I know linux likes to cache memory |
| 21:06.05 | schotten83 | jamincollins, I know that this is a poor solution and I dont know if u hacked ur code at some time, but u could try to update using the svn? |
| 21:06.14 | RoyK | dlynes_laptop: recompile asterisk with astmm enabled in the makefile |
| 21:06.22 | afrosheen | ok everyone what's that command that _really_ shows ram usage |
| 21:06.35 | dlynes_laptop | RoyK: That's what I did...that's how I know where the memory is being hogged up |
| 21:06.37 | afrosheen | I read about it last week but already forgot |
| 21:06.38 | CunningPike | dlynes_laptop: !!! |
| 21:06.39 | RoyK | dlynes_laptop: then you get 'show memory allocations' and 'show memory summary' command |
| 21:06.40 | RoyK | s |
| 21:06.41 | jamincollins | no hacked code... just a mysterious dropping of calls that I can't track down... |
| 21:06.41 | dlynes_laptop | CP!!!!!! |
| 21:06.43 | CunningPike | :D |
| 21:06.46 | schotten83 | afrosheen, free? |
| 21:07.10 | dlynes_laptop | <PROTECTED> |
| 21:07.22 | dlynes_laptop | <PROTECTED> |
| 21:07.26 | RoyK | ~pb |
| 21:07.30 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 21:07.30 | dlynes_laptop | both of those seem awfully high |
| 21:07.38 | dlynes_laptop | I was only doing two lines, RoyK |
| 21:07.41 | dlynes_laptop | not the whole dump :) |
| 21:07.44 | RoyK | sorry |
| 21:07.47 | RoyK | i thought you were doing more |
| 21:07.53 | RoyK | but none of those are really high |
| 21:07.57 | SomeJ | hehe |
| 21:08.00 | RoyK | 66188==60kB |
| 21:08.08 | dlynes_laptop | RoyK: oh...nvm |
| 21:08.24 | dlynes_laptop | RoyK: i must be drunk....i was thinking it said 468232 kilobytes :p |
| 21:08.27 | RoyK | dlynes_laptop: do a ps axv and pastebin it |
| 21:08.31 | RoyK | :) |
| 21:08.59 | dlynes_laptop | yeah..nvm...it's a linux issue |
| 21:09.07 | dlynes_laptop | stupid linux caching and buffering memory |
| 21:09.07 | RoyK | nvm? |
| 21:09.11 | dlynes_laptop | nevermind |
| 21:09.24 | jamincollins | Fetching external item into 'zaptel-1.2/oct612x' <-- that would do it |
| 21:09.32 | RoyK | dlynes_laptop: ps axfv tells you what's there |
| 21:09.47 | RoyK | you just gotta learn to use the OS |
| 21:09.48 | RoyK | :P |
| 21:10.01 | *** part/#asterisk jmls (n=asterisk@host81-159-195-120.range81-159.btcentralplus.com) |
| 21:10.22 | Zodiacal | anyone know if theres a windows printer driver that will fax via asterisk? |
| 21:10.29 | JohnRobert | is there a simple way of turning an asterisk channel into a sip channel? I have a voip provider which only uses asterisk, and i'd like to use ekiga with it...ekiga doesn't currently support aix. |
| 21:10.34 | dlynes_laptop | RoyK: I usually use ps auxffwww (it's more verbose than ps axvf) |
| 21:11.00 | dlynes_laptop | erm |
| 21:11.03 | dlynes_laptop | 2w's, not 3 |
| 21:11.58 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.42.141) |
| 21:12.01 | RoyK | iirc w doeosn't show memory status |
| 21:12.15 | RoyK | <PROTECTED> |
| 21:12.35 | JohnRobert | iax I mean |
| 21:12.45 | RoyK | yaix |
| 21:13.35 | dlynes_laptop | RoyK: you mean RSS? |
| 21:14.00 | RoyK | <PROTECTED> |
| 21:14.01 | RoyK | <PROTECTED> |
| 21:14.18 | RoyK | yeah |
| 21:14.31 | RoyK | RSS is the one you should worry about if it climbes too high |
| 21:14.35 | dlynes_laptop | ok...I don't have a 'SIZE' column |
| 21:14.40 | dlynes_laptop | only the 'RSS' column |
| 21:14.47 | RoyK | DRS, that is |
| 21:14.48 | dlynes_laptop | and that's on both ps axvf and ps auxffww |
| 21:14.49 | RoyK | iirc |
| 21:14.55 | dlynes_laptop | ah..ok |
| 21:15.03 | dlynes_laptop | drs is on ps axvf, but not ps auxffww |
| 21:15.20 | RoyK | ah |
| 21:15.30 | RoyK | it seems u implicitly gives v |
| 21:15.32 | RoyK | or something |
| 21:15.40 | RoyK | ah |
| 21:15.40 | RoyK | ok |
| 21:15.47 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.42.141) |
| 21:15.55 | RoyK | RSS is the important one anyway |
| 21:16.22 | dlynes_laptop | Yeah..I knew about RSS |
| 21:16.34 | RoyK | but still, is asterisk eating memory? |
| 21:16.38 | dlynes_laptop | %mem is important, too, if you want to see a relative measure |
| 21:16.50 | RoyK | i can see some leaks, but that's over time |
| 21:16.51 | dlynes_laptop | nah...I got confused...was thinking the allocations were in kb, not bytes |
| 21:17.04 | RoyK | it _says_ bytes :) |
| 21:17.14 | dlynes_laptop | yeah...my brain is confuzzled today :((( |
| 21:18.00 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 21:18.54 | *** join/#asterisk volp (n=volp@201.210.82.39) |
| 21:19.23 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 21:20.33 | dlynes_laptop | CunningPike: oh yeah |
| 21:20.36 | dlynes_laptop | CunningPike: btw |
| 21:20.46 | CunningPike | ? |
| 21:20.51 | dlynes_laptop | CunningPike: I'm over in Richmond regularly now (at least three times a week) |
| 21:20.52 | *** join/#asterisk ziwapandey1980 (n=ziwapand@61.246.42.141) |
| 21:21.01 | CunningPike | Cool |
| 21:21.05 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 21:21.16 | dlynes_laptop | CunningPike: yeah...met a super cute girl from Guangdong :0 |
| 21:21.34 | dlynes_laptop | CunningPike: she lives over by Richmond Center |
| 21:21.49 | mercestes | <PROTECTED> |
| 21:21.54 | mercestes | is she a geek? |
| 21:21.59 | dlynes_laptop | mercestes: not a chance |
| 21:22.04 | mercestes | aww..damn. |
| 21:22.05 | dlynes_laptop | mercestes: she's an exporter |
| 21:22.10 | mercestes | get a geeky girl. |
| 21:22.15 | dlynes_laptop | nah |
| 21:22.25 | dlynes_laptop | geeks aren't into exercise |
| 21:22.28 | mercestes | I got me an 18 year old asian chick from LA off of NeverWinter Nights...that's the way to go. |
| 21:22.53 | dlynes_laptop | gotta find a girl into exercise |
| 21:22.53 | dlynes_laptop | then she's got a nice butt :) |
| 21:23.07 | mercestes | Asians don't have to exercise to have a nice butt..they're just given one for being asian I guess...>.> |
| 21:23.09 | dlynes_laptop | besides....18 years old is a child to me :) |
| 21:23.24 | mercestes | yea, but you can raise 'em any way you want 'em that way too. |
| 21:23.57 | justdave | is there documentation anywhere for participant and moderator commands in MeetMe conference rooms? |
| 21:25.52 | justdave | most conference systems have something a participant can dial to mute themselves for example |
| 21:26.36 | justdave | I've been digging around the various asterisk sites and googling for an hour, and finding tons of stuff about how to configure the server end of meetme but nothing about participating in a conference |
| 21:27.10 | h3x | dlynes_laptop: its the genetics, not the age that counts :) |
| 21:27.11 | afrosheen | freepbx includes stuff for * commands while in a conference |
| 21:27.13 | *** join/#asterisk saftsack (n=saftsack@p54A7E515.dip.t-dialin.net) |
| 21:27.20 | afrosheen | and by * I mean 'star' :) |
| 21:28.10 | dlynes_laptop | h3x: ummm....18 is waaaaaaaaaaaaaaaaaaaaaaaay too young for this guy |
| 21:28.25 | dlynes_laptop | h3x: 18 is young enough to be my daughter for crying out loud :p |
| 21:28.33 | mercestes | *thinks* |
| 21:28.41 | mercestes | Yea, 18 is not quite young enough to be my daughter so I'm ok |
| 21:28.46 | h3x | my friend Dan married a russian girl thats 8 years younger than his youngest daughter |
| 21:29.01 | dlynes_laptop | that's just plain sick :) |
| 21:29.04 | h3x | whenever he explains this to a bartender, he gets free drinks. |
| 21:29.12 | *** join/#asterisk EvilDeshi (n=evildesh@pool-71-98-10-135.mdsnwi.dsl-w.verizon.net) |
| 21:29.12 | justdave | ok, yeah, I see the option in the config screen for conferences on freepbx for allowing access to the menu |
| 21:29.29 | [TK]D-Fender | Like RATT said "I want a woman, not some some teacher's pet ; who's trying to grow up, but hasn't made it quite yet." |
| 21:29.31 | justdave | so that's probably not turned on in the existing conferences, probably once you hit that it'll tell you what you can do in the audio help |
| 21:29.38 | h3x | I want to be like him when I grow up. |
| 21:29.49 | dlynes_laptop | but mind you |
| 21:29.57 | dlynes_laptop | this chinese girl I met is still ten years younger than me :p |
| 21:30.15 | Nugget | "still"? won't she always be? |
| 21:30.25 | dlynes_laptop | nah |
| 21:30.29 | dlynes_laptop | i get younger every year |
| 21:30.29 | h3x | not if he invents a time machine |
| 21:30.47 | dlynes_laptop | Every time I get my hair cut, I lose five years :p |
| 21:31.23 | eKo1 | You must be pretty old then. |
| 21:31.33 | dlynes_laptop | nah |
| 21:31.37 | dlynes_laptop | cunningpike is old |
| 21:31.40 | dlynes_laptop | I'm a young 'un |
| 21:32.01 | CunningPike | Whippersnapper |
| 21:32.05 | dlynes_laptop | heh |
| 21:32.29 | CunningPike | dlynes_laptop: Got my own asterisk server at home now |
| 21:32.33 | dlynes_laptop | cool |
| 21:32.40 | CunningPike | dlynes_laptop: IAX trunk back to the office |
| 21:32.53 | CunningPike | dlynes_laptop: It's running on a 6500 PowerMac lol |
| 21:32.58 | dlynes_laptop | ah...sorta like your own personal cell phone at home to tie you to your office |
| 21:33.09 | dlynes_laptop | so now you can piss your wife off even cheaper :p |
| 21:33.16 | CunningPike | lol |
| 21:33.25 | *** part/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 21:33.28 | smackus | ok, another lame question from me... when a call is delivered to an extension from a queue, is there any way to make that queue follow the dial plan of the extension? it seems like that if a call is sent to queue and from there goes to an extension, if that extension is logged in I would like it to handle the call. for example, go to voicemail, run and AGI script something like that. but it just rings for a few seconds and moves onto the next extension in |
| 21:33.32 | CunningPike | I'm going to get an SPA-3000 and put the house phone on it |
| 21:33.46 | smackus | i know that is probably how it is supposed to work, but i am looking for an alternative |
| 21:34.09 | afrosheen | smackus: that question made my brain hurt, you owe me one coffee |
| 21:34.21 | smackus | sorry |
| 21:34.59 | smackus | all i want is for it to follow the dial plan of the extension if the queue reaches that extension and is not answered, rather than moving on to the next extension logged into the queue |
| 21:35.04 | Jason99 | Is there an easy way to make asterisk control the number is sip calls through a SIP account.. inbound and outbound totalled ? |
| 21:35.30 | Jason99 | "number of" rather |
| 21:36.11 | sevard | dlynes_laptop: sup d00d |
| 21:36.17 | dlynes_laptop | CunningPike: yeah...got gurpreet's house phone running on a sipura 3000 and an asterisk box running on 64MB's of RAM and a P75 :) |
| 21:36.28 | dlynes_laptop | sevard: not much...how about you? |
| 21:36.37 | sevard | not much either, working |
| 21:36.45 | sevard | how's voicemail2 comming ? |
| 21:37.45 | SwK[Work] | ahhh thats what you are using if you are running asterisk now |
| 21:38.00 | sevard | really |
| 21:38.03 | CunningPike | smackus: It's not transferring to that extension - it's dialing the SIP UA associated with the agent or queue member. |
| 21:38.07 | SwK[Work] | yeah for a couple years now |
| 21:38.11 | sevard | you've migrated your code in the trunk already? |
| 21:38.27 | sevard | oh, what, i thought the current release was voicemail and dlynes was working on vm2 |
| 21:38.30 | CunningPike | dlynes_laptop: Yup - 64MB here too |
| 21:38.35 | afrosheen | dlynes_laptop: you win the lame hardware prize :) |
| 21:38.42 | SwK[Work] | sevard: no |
| 21:38.52 | SwK[Work] | VM2 was out 2 years ago or so |
| 21:38.57 | CunningPike | dlynes_laptop: Asterisk really does run on anything |
| 21:39.07 | sevard | dlynes: then what the hell are you working on? |
| 21:39.17 | afrosheen | there's an asterisk stack for WRAP and for the linksys routers as well correct? |
| 21:39.32 | sevard | afrosheen: there are asterisk packages for openwrt |
| 21:39.52 | dlynes_laptop | sevard: it's kinda not...been too busy finishing off our billing system, and doing some cabling jobs |
| 21:40.06 | dlynes_laptop | sevard: and i've had a bit of a side project lately doing some code for a forex project |
| 21:40.17 | afrosheen | *shiver* cabling jobs |
| 21:40.23 | dlynes_laptop | afrosheen: yeah, no doubt |
| 21:40.26 | dlynes_laptop | afrosheen: i hate cabling |
| 21:40.31 | sevard | i think everyone does |
| 21:40.34 | afrosheen | we hire guys to do that now |
| 21:40.37 | Nugget | file does work for a triple x project. |
| 21:40.53 | *** join/#asterisk bhrobinson (n=brobinso@216.85.121.26) |
| 21:41.18 | bhrobinson | hi. I cannot get my te210p to sync to my pri |
| 21:41.21 | sevard | dlynes_laptop: you ought to join Planner and fix it |
| 21:41.25 | bhrobinson | anyone able to help? |
| 21:41.32 | dlynes_laptop | sevard: planner? |
| 21:42.03 | sevard | it's a project manager application for gnome, well, in the gnome tree at least. i'm supposed to manage my own projects now while my manager does nothing |
| 21:42.06 | kink0 | bhrobinson, set your PCI clock not over 66Mhz |
| 21:42.14 | CunningPike | bhrobinson: What errors are you getting? Bear in mind your card comes with Digium installation support..... |
| 21:42.34 | bhrobinson | no errors that I can find... just no spans |
| 21:42.48 | sevard | I created a small .planner file with a very small amount of details and now i can't open it up anymore, if i open it up in planner it crashes |
| 21:42.49 | sevard | awesome |
| 21:43.13 | kink0 | bhrobinson, I have had some clocking problems with Digium cards, that was gone once I Set the PCI slot to 66Mhz |
| 21:43.13 | bhrobinson | how should the signalling be set? |
| 21:43.49 | kink0 | well, may be signalling also, but if your card works intermitently , after rebots, check for clock on the PCI slot |
| 21:44.07 | Jason99 | When a call ends does it trigger something in the context? |
| 21:44.33 | bhrobinson | kink0: will do |
| 21:44.39 | bhrobinson | trying that now |
| 21:44.39 | CunningPike | Jason99: You can use the 'h' extension for processing apres hangup |
| 21:45.00 | Jason99 | CUnningPike, will that get called no matter which end hangs up ? |
| 21:45.15 | CunningPike | Jason99: Yes |
| 21:45.21 | Jason99 | CunningPike: Thank you sir |
| 21:45.38 | *** join/#asterisk awannabe (n=brad@71-223-43-25.phnx.qwest.net) |
| 21:47.23 | smackus | <PROTECTED> |
| 21:47.31 | dlynes_laptop | sevard: no...i ought not to, then |
| 21:47.53 | dlynes_laptop | sevard: I don't have the patience to satisfy all the dependencies for anything even remotely related to Gnome |
| 21:47.53 | sevard | gay |
| 21:47.53 | CunningPike | smackus: Make what? |
| 21:48.02 | CunningPike | dlynes_laptop: rofl |
| 21:48.05 | sevard | dlynes_laptop: I use dropline gnome |
| 21:48.16 | dlynes_laptop | sevard: there's a plague on slackware if I ever saw one |
| 21:48.20 | sevard | no shit |
| 21:48.22 | bhrobinson | I get a ztspanconfig error on boot if that helps |
| 21:48.24 | sevard | that fucking thing installed PAM |
| 21:48.27 | sevard | PAM i tell you, PAM! |
| 21:48.30 | dlynes_laptop | hahahhahahaha!!!!!! |
| 21:48.33 | smackus | for example, if the queue dials extension 6000 and it does not answer, can i make it exacute a php script with the extension 6000 used as a variable? |
| 21:48.39 | sevard | piece of shit. |
| 21:48.51 | CunningPike | bhrobinson: Sounds like your card isn't installing properly. Call Digium |
| 21:48.52 | smackus | i am trying to write a script for auto log out on no answer |
| 21:48.57 | [TK]D-Fender | smackus : yes |
| 21:48.57 | bhrobinson | root |
| 21:49.21 | sevard | don't ever use dropline gnome unless you want god damned PAM |
| 21:49.22 | CunningPike | smackus: It's not dialing extension 6000 - it's calling SIP/6000 - not the same |
| 21:49.31 | bhrobinson | it was working at my office... I just moved it to a different |
| 21:49.34 | smackus | right... thats what I mean. |
| 21:49.47 | dlynes_laptop | sevard: don't ever use dropline |
| 21:49.49 | dlynes_laptop | sevard: period |
| 21:49.55 | smackus | [TK]D-Fender: lead me |
| 21:50.05 | sevard | it looks like it was written by a bunch of 12 year olds |
| 21:50.06 | smackus | what to read? |
| 21:50.20 | smackus | i got the chanisavailable stuff working, so this is my last thing to figure out |
| 21:50.40 | CunningPike | smackus: Why are you in a queue if you don't want to go to the next available agent? |
| 21:51.40 | smackus | I am trying to make the autologout script a little cleaner that what I have seen with the set up now. i have huge config files to be able to do all of this agent stuff using vmauthenticate. |
| 21:52.21 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 21:53.10 | *** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com) |
| 21:53.21 | BlepsoaF | hello all, has anyone used a cisco 7941? |
| 21:53.31 | smackus | want to make it cleaner |
| 21:54.37 | smackus | [TK]D-Fender: essentially what I am looking to do is if a queued call is delivered to SIP/6000 and that call is not answered, I want to execute a php script which logs out the extension SIP/6000 and then put the caller back into queue |
| 21:54.55 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 21:54.56 | dserban | will the asterisk queue system deliver calls sequentially and concurrently? |
| 21:55.03 | CunningPike | smackus: There's a setting for that....... |
| 21:55.07 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 21:55.10 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 21:55.12 | dserban | ie... there are 4 calls in the queue and ther eare 4 agents available... |
| 21:55.36 | *** join/#asterisk joako (n=kvirc@64.238.175.93) |
| 21:55.40 | dserban | the queue system tries to offload caller 1 and then 2 and then 3... can I make asterisk ring all 4 available interfaces at the same time (theoretically?) |
| 21:55.46 | smackus | CunningPike: are you referring to the agents.conf file? isnt that depricated? I am using addqueuemember and removequeuemember. |
| 21:55.55 | smackus | is there a way to still use the agents.conf with those commands? |
| 21:56.07 | dserban | if you don't wanna type.. point me to the voip-info.org page :) |
| 21:56.19 | Qwell | dserban: there was an autofill option added a while back |
| 21:56.33 | dserban | Qwell: thanks i will look it up. |
| 21:56.35 | dserban | appreciated |
| 21:56.42 | Qwell | I think it was only trunk though |
| 21:56.47 | BlepsoaF | if anyone has used a 7941 cisco phone, could they send me their cnf.xml file? I tried the one off the wiki but its not working for some reason :S |
| 21:57.04 | dserban | Oh i see. |
| 21:57.06 | CunningPike | smackus: Hmmm - don't think so........ |
| 21:57.13 | dserban | so not in the current released version? |
| 21:57.20 | Qwell | dserban: not afaik |
| 21:57.23 | CunningPike | smackus: I think autologoff is in agents.conf |
| 21:57.24 | dserban | shit.. |
| 21:57.26 | *** join/#asterisk rollergrrl (n=0x3e44d@71-213-6-123.slkc.qwest.net) |
| 21:57.37 | smackus | it is, but i dont think that AddQueueMember can use that file |
| 21:57.41 | rollergrrl | Is there a way to see debug messages for rxfax? |
| 21:57.46 | smackus | I would love to learn otherwise. |
| 21:57.48 | dlynes_laptop | any news on when 1.4 is coming out, if it's ever coming out? |
| 21:57.56 | Qwell | dlynes_laptop: real soon now |
| 21:58.00 | dlynes_laptop | heh |
| 21:58.01 | ppdm | anyone use a cisco 7961? |
| 21:58.49 | BlepsoaF | ppdm: I have a cisco 7941 |
| 21:59.09 | CunningPike | autologoff should really be a queues.conf setting - Qwell, fix that would ya? ;) |
| 21:59.19 | Qwell | why would it be? |
| 21:59.24 | smackus | is it true that agentcallbacklogin is not going to be fixed and removed from the next version? |
| 21:59.29 | dserban | so trunk is the real soon now (tm) version of asterisk? |
| 21:59.36 | *** join/#asterisk EvilDeshi (n=rattfink@pool-71-98-10-135.mdsnwi.dsl-w.verizon.net) |
| 21:59.38 | Qwell | dserban: pretty much |
| 21:59.54 | CunningPike | Qwell: So those who use dynamic members can still log them out of they don't answer |
| 22:00.05 | CunningPike | s/ of / if / |
| 22:00.06 | Qwell | smackus: It may or may not be fixed. Yes, it will be removed in 1.6 |
| 22:00.15 | dserban | Meh... Any suggestions if I'm running 1.2? I don't want to make 100 callers theoretically wait 3000 seconds potentially. |
| 22:00.17 | ppdm | BlepsoaF: my 7961 wont re-read the changes on the config file |
| 22:00.21 | ppdm | via TFTP |
| 22:00.23 | Qwell | CunningPike: There are other methods for doing that |
| 22:00.26 | Qwell | like...autopause |
| 22:00.27 | ppdm | have you had any luck with that? |
| 22:00.31 | Qwell | or whatever it was |
| 22:00.31 | EvilDeshi | well i got all the ports forwarded and my soft phones register but i still dont get any audio passing through i can hear the IVM just fine and if i call with a regular phone i can hear audio through the phone but i cant hear audio on the soft phones |
| 22:00.44 | Qwell | yes, autopause |
| 22:00.49 | EvilDeshi | IVR* |
| 22:01.11 | CunningPike | Qwell: Trunk? |
| 22:01.15 | Qwell | yes |
| 22:01.23 | dserban | I should just probably lower the ring time. And that'll probably buy us some time until we can test 1.4... |
| 22:01.26 | dserban | how stable is trunk/ |
| 22:01.27 | dserban | ? |
| 22:01.29 | CunningPike | Qwell: So, when is 1.4 out again? ;) |
| 22:01.36 | *** join/#asterisk Lyfe (n=lyfe@69.8.146.78) |
| 22:01.40 | *** part/#asterisk Qwell (i=north@unaffiliated/qwell) |
| 22:01.40 | *** join/#asterisk shap (n=shap@c-68-33-84-43.hsd1.md.comcast.net) |
| 22:01.46 | *** join/#asterisk Qwell (i=north@unaffiliated/qwell) |
| 22:01.46 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:01.49 | Qwell | woops |
| 22:01.53 | Qwell | <Qwell> real soon now |
| 22:01.58 | awannabe | for doing a frac PRI and data on the same PRI is a Sangoma or a Diigum the best way to fly? I will be using a external firewall, so the * box will not be the firewall as well |
| 22:01.59 | CunningPike | lol - thought I'd pissed you off |
| 22:02.00 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 22:02.10 | CunningPike | It's Real Soon Now(tm) |
| 22:04.09 | vader-- | anyone upgraded to 1.2.12 and 1.2.9 zaptel yet? |
| 22:04.34 | *** part/#asterisk afrosheen (n=test@txprotoa2.august.net) |
| 22:04.38 | shap | vader: I'm on 1.2.12, but not zaptel |
| 22:04.57 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 22:04.57 | vader-- | im looking to go from 1.2.7 but im kinda scared |
| 22:05.04 | vader-- | because it's a production system |
| 22:05.15 | vader-- | and it's running stable im just having a voicemail problem |
| 22:05.21 | bhrobinson | ok, found my issue |
| 22:05.37 | shap | vader: good reason to be scared :-) Put in a test system in parallel? |
| 22:05.39 | bhrobinson | on the followme feature, how do I need to put in the number? |
| 22:05.57 | shap | all: I'm having a config puzzle. |
| 22:06.15 | joako | Anyone ever get a UTStarcom phone to work with asterisk? |
| 22:06.35 | Lyfe | can anyone tell me how zaptel & asterisk would handle a dchan on a PRI with a sangoma card? I've been told by sangoma that their FreeBSD driver does not support hardware d-chan, but have no idea how to make the zaptel drivers and/or asterisk handle the d-chan. |
| 22:07.49 | Corydon-w | I'm not aware that the zaptel drivers were ever successfully ported to FreeBSD for the T1 cards |
| 22:07.53 | shap | I have en extension "500". For testing, I'm wiring it through to ekiga.net. When the Dial() line is SIP/500@ekiga.net, all is well. When I change the Dial() line to SIP/500@ekiga {which is the name of my sip config for the peer), I get an authentication error. |
| 22:08.04 | bkw_ | Corydon-w: you must have been sleeping |
| 22:08.18 | Lyfe | Corydon-w: well, they definitely work with t1 wink circuits. |
| 22:08.19 | *** join/#asterisk Mportnoy (n=test@200.122.158.88) |
| 22:08.25 | awannabe | so is Digium or Sangoma better for a voice/data combo setup? |
| 22:08.30 | Corydon-w | bkw_: good, you can answer Lyfe's question |
| 22:08.31 | bkw_ | Sangoma |
| 22:08.32 | shap | <PROTECTED> |
| 22:08.37 | Corydon-w | awannabe: Digium |
| 22:08.38 | dlynes_laptop | awannabe: they're both fine |
| 22:08.44 | mog | awannabe, they are about the same |
| 22:08.46 | awannabe | which one is easier to setup? |
| 22:08.50 | bkw_ | no Digium isn't "fine" its "ok" |
| 22:08.52 | shap | I'm sure this is a FAQ, so I'ld be happy with a pointer if somebody has one, but I'm stumped. |
| 22:08.53 | bkw_ | but I wouldn't call it fine |
| 22:08.56 | Corydon-w | Digium is easier to set up |
| 22:09.01 | bkw_ | wrong again |
| 22:09.03 | Lyfe | and from the description from the Sangoma Software proj leader for FreeBSD drivers, it's definitely possible. |
| 22:09.16 | bkw_ | its just as easy to setup |
| 22:09.20 | awannabe | both can hand the data off to another device on the LAN right? |
| 22:10.07 | Mportnoy | Hello I have my iptables log configure to LOG , but my console screen is not usable with all that info on display how can modify that? |
| 22:10.11 | Corydon-w | awannabe: you pretty much have to handle the data at that point |
| 22:10.23 | awannabe | ok |
| 22:10.46 | Corydon-w | awannabe: remember that you're decoding the HDLC frames, so unless you're handing the frames off to another T1 circuit, that's where the endpoint is |
| 22:10.51 | smackus | Qwell: so is an alternative that is not quite as bloated as the vmauthenticate method going to be added to replace the agentcallbacklogin? or a more simple and clean way to use autologoff? |
| 22:11.19 | Qwell | bloated? |
| 22:11.29 | awannabe | Corydon-w, yeah, but cant you send the data somewhere else? since in this case this machine wont be the gateway/firewall for the LAN |
| 22:11.40 | smackus | well... i really have to have a lot of script in there to make that work for 100 agents |
| 22:11.58 | smackus | would be nice to just toggle it per queue |
| 22:12.16 | Corydon-w | awannabe: well, it could be the gateway for the DMZ and use a different machine to gateway to the LAN. That's how our network is setup |
| 22:12.34 | smackus | addqueuemember works fine for me, but I am really missing the autologgoff feature |
| 22:12.42 | niter3 | hey guys.. I want to setup an extension that will pass 1-areacode if a user just dials 7 digits |
| 22:12.45 | niter3 | how can I do this? |
| 22:12.47 | awannabe | Corydon-w, ok, so then you just assign a IP to the sangoma adapter basically, and then its the "true" gateway then, right? |
| 22:13.09 | Corydon-w | awannabe: it's possible, but it's not the optimal configuration |
| 22:13.16 | Qwell | smackus: autopause |
| 22:13.17 | CunningPike | niter3: Dial(whatever/1604${EXTEN}) |
| 22:13.33 | Corydon-w | awannabe: what we do is to use fake IPs for the T1 as a PTP tunnel |
| 22:13.34 | niter3 | duh ... |
| 22:13.36 | niter3 | ok cool |
| 22:13.39 | joako | If I set in sip.conf the port to 5060, I can use that and which other ports in the client? |
| 22:13.40 | smackus | interesting... so that will pause them, but leave them logged in? |
| 22:13.57 | Corydon-w | awannabe: that way, you don't lose the use of any IPs in your allocation |
| 22:14.08 | awannabe | Corydon-w, ahh ok, the true firewall/gateway is gonna be a crappy sonicwall, so ill guess il h ave to forward em out or something |
| 22:14.19 | hmmhesays | sonicwalls can be a pain in the ass |
| 22:14.23 | *** join/#asterisk Defraz (n=t0tal@001-775-788.area1.spcsdns.net) |
| 22:14.28 | hmmhesays | i know this well |
| 22:14.29 | awannabe | yeah, its not by choice |
| 22:14.35 | Corydon-w | Right, so you have a DMZ between the sonicwall and the t1 gateway |
| 22:14.36 | awannabe | customer already has the firggin equipment, heh |
| 22:14.42 | awannabe | yeah, ill just do that |
| 22:14.57 | awannabe | ill figure the crap out somehow, heh |
| 22:15.01 | Corydon-w | and then anything needing a real IP gets plugged in there |
| 22:15.22 | awannabe | yeah just the customers POS exchange server |
| 22:15.28 | Corydon-w | Big NOTE: you have to use the ip(8) command for setting up your routing if you use a PTP tunnel |
| 22:15.51 | awannabe | well i cant use a PTP tunnel, i dont think the sonicwalls support that at all |
| 22:15.56 | Corydon-w | It's mainly so you can get the source address of packets originating on the T1 gateway |
| 22:16.11 | Corydon-w | You're misunderstanding |
| 22:16.33 | CunningPike | To use followme for PSTN calls, you would need 2 lines, right? |
| 22:16.35 | Corydon-w | Not PPTP. A point-to-point tunnel from the T1 to the far end |
| 22:17.05 | awannabe | oh, nevermind, lol |
| 22:17.11 | awannabe | sorry im out of it |
| 22:17.12 | smackus | Qwell: looking for documentation for autopause. not finding it in the wiki... can you help? |
| 22:17.31 | Qwell | configs/queues.conf.sample |
| 22:17.40 | smackus | ok, thanks |
| 22:20.27 | awannabe | not seeing much on the wiki about same ple configs or anything for the sangoma cards |
| 22:23.50 | niter3 | CunningPike: exten => _9NXXXXXX.,1,Dial(SIP/1613${EXTEN:1}) would work? |
| 22:24.14 | dserban | CunningPike: :o LOWER MAINLAND!!11 |
| 22:24.23 | CunningPike | niter3: Should do - try it :D |
| 22:24.30 | CunningPike | dserban: You too? |
| 22:24.48 | niter3 | works |
| 22:24.51 | niter3 | i had that '.' |
| 22:24.54 | niter3 | had to remove it |
| 22:24.55 | niter3 | then it worked |
| 22:25.08 | *** join/#asterisk ibob63 (n=hp@bb-87-82-12-166.ukonline.co.uk) |
| 22:27.54 | *** join/#asterisk RoyK (n=roy@ti211210a080-0574.bb.online.no) |
| 22:30.34 | EvilDeshi | well i am at a loss with this i have gone through several tutorials on setting this up everything seems to work ivr has audio but no audio passing through on soft phones |
| 22:30.59 | niter3 | CunningPike: Do you have any idea how I could get a dial tone working after a user presses 9 so they knwo to dial out? |
| 22:31.21 | CunningPike | niter3: It's called DISA - search for that in the wiki |
| 22:31.29 | CunningPike | ~thewiki |
| 22:31.37 | jbot | hmm... thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 22:31.54 | niter3 | thanks |
| 22:32.23 | CunningPike | EvilDeshi: Codec issue, perhaps? Or NAT? |
| 22:32.40 | EvilDeshi | well i have all the ports mapped properly |
| 22:32.54 | EvilDeshi | the only thing i havent tried is messing with codecs or setting it directly on the net |
| 22:32.59 | EvilDeshi | im using ulaw alaw and gsm |
| 22:33.48 | niter3 | CunningPike: I must have wored taht wrong...... That is a cool feature but I mean anyone that is on my PBX in the company. I want it so when they hit 9 then dail a phone number out of the pbx to have a dial tone after hitting 9. |
| 22:34.59 | smackus | Qwell: So autopause is not available in 1.2.9.1? |
| 22:35.26 | EvilDeshi | now if i connect to meetme with my softphone and also calling in with a regular phone the regular phone can hear the softphone but the softphone hears nothing |
| 22:35.41 | CunningPike | niter3: I thought that DISA was used for that, as well (we're in 10-digit land, so I've not configured it at our place) |
| 22:35.50 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-92-128.cybersurf.com) |
| 22:36.16 | dserban | CunningPike: nah, okanagan, but close 'nuff ;) |
| 22:36.22 | niter3 | oh.. from what I understood it's so people outside the pbx can dialin and get a dial tone to dial out.. |
| 22:36.23 | CunningPike | EvilDeshi: I'm thinking codec issues, if you're certain you have NAT issues covered off |
| 22:36.29 | niter3 | I'm looking for a comfort tone |
| 22:36.33 | CunningPike | dserban: Nice up there |
| 22:36.35 | niter3 | so people know when to dial out |
| 22:36.41 | dserban | Yeah except for the smoke :( |
| 22:36.47 | EvilDeshi | im pretty certian on the nat issues ive forwarded every port only to the asterisk box |
| 22:37.00 | EvilDeshi | and external clients can register their phones fine and access voicemail |
| 22:37.10 | EvilDeshi | but call to call and meetme isnt working right |
| 22:37.35 | CunningPike | EvilDeshi: Which softphone? |
| 22:38.03 | EvilDeshi | eyebeam |
| 22:40.31 | dserban | Asked to transmit frame type 2, while native formats is 4 (read/write = 4/2) <--- what's that caused by? |
| 22:41.28 | CunningPike | dserban: Codec mismatch |
| 22:41.54 | dserban | Hrrm, and how did I manage to do that? :( |
| 22:42.07 | dserban | gsm throughout.. |
| 22:42.14 | dserban | rather, can I find out what 4 and 2 are? |
| 22:42.34 | C6Vette | 6? |
| 22:42.40 | bkw_ | no |
| 22:43.02 | Corydon-w | dserban: show codec 6 |
| 22:43.17 | dserban | ahh.. |
| 22:43.47 | bkw_ | ulaw vs gsm |
| 22:43.53 | bkw_ | 6 isn't valid |
| 22:44.11 | EvilDeshi | im gonna throw in the towel on this one |
| 22:44.15 | bkw_ | well 6 is a mix of gsm/ulaw |
| 22:44.43 | bkw_ | anyway |
| 22:44.46 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 22:44.47 | *** join/#asterisk zeppelin_ (n=zeppelin@201.66.208.174) |
| 22:48.38 | *** join/#asterisk budairc (n=chatzill@200.215.57.174) |
| 22:49.17 | [TK]D-Fender | niter3 : And I explained earlier taht its up to your ATA to provide that extra tone and none that I know of will. |
| 22:49.22 | *** join/#asterisk m_a_g_o (i=maxgluck@201.243.103.240) |
| 22:52.17 | *** part/#asterisk ibob63 (n=hp@bb-87-82-12-166.ukonline.co.uk) |
| 22:54.15 | m_a_g_o | good evening... I'm trying to send an e-mail from within an agi script, and everything seems to be working all right... the program (sendmail) is opened, the email is printed and it seems to be closed as the '||' clauses after open, print and close are not being called. does it have to do something with AGIs not being able to print to output as it happens with STDERR? |
| 22:54.35 | m_a_g_o | I appreciate any thogh, thanks! |
| 22:54.44 | m_a_g_o | *thought |
| 22:57.37 | CunningPike | niter3, [TK]D-Fender: The Sipuras will - "An “outside line” dial tone can be generated within a sequence by appending a ‘,’ character between digits. Thus, the sequence “9, 1xxxxxxxxxx” sounds an “outside line” dial tone after the user presses ‘9’, until the ‘1’ is pressed. |
| 23:01.57 | m_a_g_o | np, working now, thx |
| 23:02.22 | m_a_g_o | if anyone is interested... abuse control and clid reporting by e-mail from astcc.agi |
| 23:02.30 | m_a_g_o | byeee! |
| 23:02.47 | m_a_g_o | ;) |
| 23:02.47 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:08.14 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 23:09.28 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 23:14.56 | type0 | does asterisk business edition list which dialogic cards they support? |
| 23:16.55 | *** part/#asterisk bkruse (i=bkruse@nat/digium/x-75020f7d3210c469) |
| 23:22.03 | denon | what happened to directory? |
| 23:22.13 | Zodiacal | does ocr software work ok on faxs or is the quality to bad? |
| 23:22.28 | file | denon: gone... GONE! |
| 23:22.37 | denon | good riddance, I say |
| 23:22.39 | Nugget | it's easier to parse a fax than is to parse you on irc. :P |
| 23:22.40 | denon | he didnt talk much anyway |
| 23:23.03 | Zodiacal | nugget :/ |
| 23:23.21 | Nugget | file should have just grabbed a plane right from downtown montreal. |
| 23:23.36 | Nugget | http://discussions.flightaware.com/viewtopic.php?p=15400 |
| 23:23.44 | file | geez finding the right place to go to grab a US connection was hard |
| 23:25.09 | file | the lag on this wifi network is horrible |
| 23:25.46 | file | 900ms |
| 23:27.54 | [TK]D-Fender | file : You are? |
| 23:28.07 | [TK]D-Fender | file : on your way elsewhere? |
| 23:28.18 | file | [TK]D-Fender: going to Boston for Fall VON |
| 23:28.30 | [TK]D-Fender | Ah..... |
| 23:28.46 | file | I'm playing a game of "guess who else is going to VON based on what they look like and other characteristics" |
| 23:29.09 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:29.15 | file | I've picked out 3 people so far |
| 23:30.54 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:30.54 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:30.59 | file | hey mog |
| 23:31.51 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 23:31.51 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 23:33.06 | *** join/#asterisk tessier_ (n=treed@gw.drjays.com) |
| 23:33.57 | tessier_ | Anyone using a headset with a snom phone? Are they just standard headsets? We have two models of headsets. Both work with our old phone system. But on the snom phones one kind of headset works and the other does not. |
| 23:34.45 | [hC] | im still trying to figure out how to retrofit my polycom phones to make my plantronics hl10 lifter work right |
| 23:34.50 | [hC] | the handset always falls |
| 23:34.52 | [hC] | or shifts too far |
| 23:36.16 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 23:36.42 | Dr-Linux | hi guys |
| 23:37.24 | mog | hey file |
| 23:37.35 | file | mog: how goes? |
| 23:37.47 | Dr-Linux | what would be new features in version 1.4 ? |
| 23:38.01 | mog | teleportation |
| 23:38.15 | file | transmutation |
| 23:38.31 | Qwell | teletransmutation |
| 23:38.34 | Dr-Linux | mog: anywhere i can see the 1.4 feature details? |
| 23:38.59 | file | Dr-Linux: there's some on Kevin's interview on sineapps |
| 23:39.02 | mog | ooh |
| 23:39.04 | [hC] | anyone ever try broadcasting video to a cisco 7970? |
| 23:39.09 | [hC] | or know if its possible today? |
| 23:39.11 | file | but no formal list has yet to have been released |
| 23:39.16 | Qwell | [hC]: It isn't possible. |
| 23:39.20 | mog | the only one that matters is jabber ^_^ or vldtmf i cant remember which |
| 23:39.29 | [hC] | Qwell: because we dont know how to do video with SCCP yet? |
| 23:39.30 | Qwell | You *can* use a meta refresh, to send images to it at some interval |
| 23:39.33 | Dr-Linux | file: i tried on google, but couudn't find can you please give the link? |
| 23:39.34 | file | mog: both! |
| 23:39.38 | Qwell | [hC]: because the xml services doesn't support it |
| 23:39.53 | Qwell | that phone isn't powerful enough for any video |
| 23:39.57 | [hC] | Qwell: it displays a video camera icon in the corner, implying it can do full motion video calls |
| 23:40.04 | Qwell | It can. |
| 23:40.08 | Qwell | through your PC ;) |
| 23:40.11 | [hC] | Qwell: and im pretty sure there is a video conferencing camera available. |
| 23:40.12 | Qwell | Cisco VT Advantage |
| 23:40.14 | Dr-Linux | [hC]: SCCP is not a part of asterisk |
| 23:40.17 | file | Dr-Linux: http://www.sineapps.com/news.php?rssid=1483 |
| 23:40.19 | Qwell | [hC]: google that |
| 23:40.26 | [hC] | Will do:) |
| 23:40.31 | Qwell | it's a hack |
| 23:40.35 | [hC] | I have a customer who has a video enabled door entry system |
| 23:40.40 | file | I think we're doing pre boarding |
| 23:40.46 | [hC] | they want to be able to receive a call and see who's at the door |
| 23:40.50 | [hC] | even if its not full framerate |
| 23:40.54 | file | gotta go |
| 23:40.56 | file | ttyl |
| 23:41.00 | Dr-Linux | Qwell: you should conftribute SCCP for asterisk |
| 23:41.15 | Qwell | Dr-Linux: it isn't mine |
| 23:41.19 | Qwell | file: pfft |
| 23:41.24 | Qwell | file: They can wait for you |
| 23:41.32 | Qwell | say "I'm not done talking on IRC yet, nubs" |
| 23:41.50 | Dr-Linux | Qwell: who's gonna built sccp for * ? |
| 23:42.00 | [hC] | Qwell: so, this VT advantage doesnt actually display the video on the phone itself, its done on the pc's display |
| 23:42.06 | Qwell | exactly |
| 23:42.12 | Qwell | [hC]: and the camera is usb |
| 23:42.29 | [hC] | right |
| 23:42.35 | [hC] | tha wont work, they dont want to have to put a pc there. |
| 23:42.37 | [hC] | :( |
| 23:42.45 | Qwell | so, this program connects through the phone (note, you MUST have your PC on the switch port, for some stupid reason), and sends the video in a different stream, iirc |
| 23:42.51 | Qwell | maybe the same stream...dunno |
| 23:43.01 | Qwell | [hC]: Get a grandstream video phone |
| 23:43.03 | [hC] | oo I found a link for a 1frame/sec image.. that might work. |
| 23:43.04 | Qwell | OR |
| 23:43.16 | Qwell | get them to buy me a 7985, and I'll get it working with chan_skinny :) |
| 23:44.39 | [hC] | Qwell: haha :) so this image refresh shit, is it just standard meta refresh or do you need to use special cisco xml? |
| 23:44.54 | Qwell | special cisco xml, with a standard meta refresh :D |
| 23:45.09 | Qwell | it's all documented |
| 23:45.18 | *** join/#asterisk ATGeek (n=atg@pdpc/supporter/student/ATravelingGeek) |
| 23:45.20 | Qwell | and, don't forget, you *CAN* open an rtp stream on the phone, via the xml |
| 23:45.38 | Qwell | so, sound can be synced, or played through the xml services |
| 23:47.08 | niZon | cisco's xml rocks |
| 23:47.19 | [hC] | the image refresh, can it be an image stream, or do you have to modify the xml file every refresh and put in encoded video? |
| 23:47.20 | Qwell | niZon: Thank you :p |
| 23:47.24 | niZon | I have a winamp remote for my 7970 |
| 23:47.32 | [hC] | Im gonna take a look for the docs on how i could do this |
| 23:47.36 | Qwell | [hC]: I don't think it displays anything like gif |
| 23:47.46 | [hC] | Is it @ cisco or voip-info? |
| 23:47.52 | niZon | it can do PNGs.. and thats about it |
| 23:47.55 | Qwell | cisco.com has it fairly in depth |
| 23:47.59 | Qwell | niZon: animated png? |
| 23:48.07 | type0 | whats a good free web based config for asterisk? |
| 23:48.15 | niZon | didn't know pngs could be animated.. |
| 23:48.17 | Qwell | type0: phpshell + vi |
| 23:48.21 | Qwell | niZon: mng or something |
| 23:48.29 | niZon | hmm |
| 23:48.48 | *** join/#asterisk compu73rg33k (n=compu73r@d53-64-54-188.nap.wideopenwest.com) |
| 23:48.49 | [hC] | so I guess what you'd do is constantly re-write the xml file and have the phone meta refresh it.. take the stream, convert the video into the encoded string, and let it take care of itself. |
| 23:49.23 | niZon | [hC]: trying to do a video phone thing? |
| 23:49.27 | Qwell | well, the 7970 can use image paths |
| 23:49.35 | Qwell | don't need to embed it |
| 23:49.47 | niZon | in my 7970 I have a video option, i have no idea how to use it though |
| 23:49.57 | Qwell | niZon: google vt advantage |
| 23:50.01 | [hC] | niZon: yeah, view security camera (for door entry) on the phone |
| 23:50.03 | Qwell | it's silly |
| 23:50.14 | [hC] | it pretty much defeats the point |
| 23:50.30 | niZon | [hC]: well for a door cam you pretty much just need a single capture.. |
| 23:50.44 | Qwell | shame you can't change the display via the sccp protocol |
| 23:50.49 | niZon | yeah |
| 23:50.50 | Qwell | That'd be real slick |
| 23:51.10 | *** join/#asterisk file2 (n=IrcNet@out.clearnet.com) |
| 23:51.10 | *** mode/#asterisk [+o file2] by ChanServ |
| 23:51.16 | Qwell | file2: nub |
| 23:51.20 | Qwell | Did you already land?! |
| 23:51.30 | file2 | greetins from an on the ground canadair plane |
| 23:51.40 | Qwell | fun |
| 23:51.41 | hmmhesays | where you headed file? |
| 23:51.44 | Qwell | grounded, are we? |
| 23:51.46 | file2 | boston. |
| 23:51.51 | Qwell | already? |
| 23:51.57 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 23:51.58 | hmmhesays | i see |
| 23:52.00 | wunderkin | file needs to load up on some beans for his muffins |
| 23:52.00 | Qwell | Didn't you just leave like...10 minutes ago? |
| 23:52.01 | mick_linux | and the wifi isn't fcking w/ the electronics? ;) |
| 23:52.10 | hmmhesays | sounds fun |
| 23:52.31 | Qwell | mick_linux: I'm starting to think that's why the plane can't/won't take off :D |
| 23:52.36 | file2 | sort of |
| 23:52.44 | mick_linux | Qwell, turn wifi OFF |
| 23:52.55 | CunningPike | file2: See any black helicopters? |
| 23:53.01 | mick_linux | for your safety and the safety of others |
| 23:53.13 | Qwell | CunningPike: He's in canada. They don't have any |
| 23:53.22 | CunningPike | Qwell: That's what you think..... |
| 23:53.31 | mick_linux | oops, wrong person |
| 23:53.35 | mick_linux | file2, ^^^ |
| 23:53.41 | mick_linux | i'm tired and confused |
| 23:54.04 | file2 | lol |
| 23:54.14 | file2 | gotta go |
| 23:54.18 | mick_linux | canada... isn't that where that WoW guy got screwed for flushing his ipod down the toilet? |
| 23:55.25 | CunningPike | mick_linux: Chicago to Ottawa |
| 23:55.26 | [hC] | niZon: yeah, im just trying to figure out how to use the xml to display a png that was retrieved remotely |
| 23:55.44 | mick_linux | that was hilarious :) |
| 23:55.50 | mick_linux | though f---ed up |
| 23:55.56 | mick_linux | still funny |
| 23:56.31 | CunningPike | mick_linux: Now that we have George W. Harper as Pres^H^H^H^HPrime Minister, we're doomed to the same "security" folderoll as everyone else |
| 23:56.41 | mick_linux | haha |
| 23:57.16 | mick_linux | i wanted to move to Canada actually, seems that people are more sain^H^H^H^Hliberal there |
| 23:57.33 | *** join/#asterisk nettie (n=nettie@85-18-54-38.ip.fastwebnet.it) |
| 23:58.00 | CunningPike | Someone needs to make an exploding password and watch the security folks' heads explode as they try and work around that problem |
| 23:58.06 | CunningPike | s/password/passport/ |
| 23:58.19 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 23:58.38 | CunningPike | "We have to ban passports - - - boom!" |
| 23:58.52 | mick_linux | CunningPike, RFID passports :-P |
| 23:59.05 | mick_linux | that way anyone can detinate it ;) |
| 23:59.55 | mick_linux | though, whoever thought that RFID would increse security needs to have their head thouroughly examined |